Merging upstream version 127.0.

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

36 files changed, 825 insertions, 440 deletions

diff --git a/third_party/libwebrtc/test/mock_frame_transformer.h b/third_party/libwebrtc/api/test/mock_frame_transformer.h
index 617cda8a43..8f438bdf9e 100644
--- a/third_party/libwebrtc/test/mock_frame_transformer.h
+++ b/third_party/libwebrtc/api/test/mock_frame_transformer.h
@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef TEST_MOCK_FRAME_TRANSFORMER_H_
-#define TEST_MOCK_FRAME_TRANSFORMER_H_
+#ifndef API_TEST_MOCK_FRAME_TRANSFORMER_H_
+#define API_TEST_MOCK_FRAME_TRANSFORMER_H_
 
 #include <memory>
 #include <vector>
@@ -42,4 +42,4 @@ class MockFrameTransformer : public FrameTransformerInterface {
 
 }  // namespace webrtc
 
-#endif  // TEST_MOCK_FRAME_TRANSFORMER_H_
+#endif  // API_TEST_MOCK_FRAME_TRANSFORMER_H_
diff --git a/third_party/libwebrtc/test/BUILD.gn b/third_party/libwebrtc/test/BUILD.gn
index 75d8d9f3a8..bf98ec7d3a 100644
--- a/third_party/libwebrtc/test/BUILD.gn
+++ b/third_party/libwebrtc/test/BUILD.gn
@@ -735,6 +735,8 @@ if (rtc_include_tests) {
         "../api:mock_video_encoder",
         "../api:scoped_refptr",
         "../api:simulcast_test_fixture_api",
+        "../api/environment",
+        "../api/environment:environment_factory",
         "../api/task_queue",
         "../api/task_queue:task_queue_test",
         "../api/test/video:function_video_factory",
@@ -867,6 +869,7 @@ rtc_library("fileutils") {
     ":fileutils_override_api",
     ":fileutils_override_impl",
     "../rtc_base:checks",
+    "../rtc_base:ssl",
     "../rtc_base:stringutils",
   ]
   absl_deps = [
@@ -956,6 +959,7 @@ rtc_library("fileutils_unittests") {
     ":fileutils",
     ":test_support",
     "../rtc_base:checks",
+    "../rtc_base:ssl",
   ]
   absl_deps = [
     "//third_party/abseil-cpp/absl/strings:strings",
@@ -1107,28 +1111,6 @@ rtc_source_set("test_renderer") {
   }
 }
 
-rtc_library("mock_frame_transformer") {
-  visibility = [ "*" ]
-  testonly = true
-  sources = [ "mock_frame_transformer.h" ]
-  deps = [
-    "../api:frame_transformer_interface",
-    "../test:test_support",
-  ]
-}
-
-rtc_library("mock_transformable_frame") {
-  visibility = [ "*" ]
-
-  testonly = true
-  sources = [ "mock_transformable_frame.h" ]
-
-  deps = [
-    "../api:frame_transformer_interface",
-    "../test:test_support",
-  ]
-}
-
 if (is_mac) {
   rtc_library("test_renderer_objc") {
     testonly = true
diff --git a/third_party/libwebrtc/test/call_test.cc b/third_party/libwebrtc/test/call_test.cc
index 6cdd8da133..f26a44a341 100644
--- a/third_party/libwebrtc/test/call_test.cc
+++ b/third_party/libwebrtc/test/call_test.cc
@@ -657,9 +657,7 @@ void CallTest::StartVideoSources() {
 void CallTest::StartVideoStreams() {
   StartVideoSources();
   for (size_t i = 0; i < video_send_streams_.size(); ++i) {
-    std::vector<bool> active_rtp_streams(
-        video_send_configs_[i].rtp.ssrcs.size(), true);
-    video_send_streams_[i]->StartPerRtpStream(active_rtp_streams);
+    video_send_streams_[i]->Start();
   }
   for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
     video_recv_stream->Start();
diff --git a/third_party/libwebrtc/test/fuzzers/BUILD.gn b/third_party/libwebrtc/test/fuzzers/BUILD.gn
index 083c20c6f4..642b0c8e08 100644
--- a/third_party/libwebrtc/test/fuzzers/BUILD.gn
+++ b/third_party/libwebrtc/test/fuzzers/BUILD.gn
@@ -132,6 +132,11 @@ if (rtc_use_h265) {
       "../../modules/video_coding/",
     ]
   }
+
+  webrtc_fuzzer_test("h265_depacketizer_fuzzer") {
+    sources = [ "h265_depacketizer_fuzzer.cc" ]
+    deps = [ "../../modules/rtp_rtcp" ]
+  }
 }
 
 webrtc_fuzzer_test("forward_error_correction_fuzzer") {
@@ -471,6 +476,7 @@ webrtc_fuzzer_test("stun_validator_fuzzer") {
 webrtc_fuzzer_test("pseudotcp_parser_fuzzer") {
   sources = [ "pseudotcp_parser_fuzzer.cc" ]
   deps = [
+    "../../p2p:pseudo_tcp",
     "../../p2p:rtc_p2p",
     "../../rtc_base:threading",
   ]
diff --git a/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..00025ef887
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/h265_depacketizer_fuzzer.cc
@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h265.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 200000)
+    return;
+  VideoRtpDepacketizerH265 depacketizer;
+  depacketizer.Parse(rtc::CopyOnWriteBuffer(data, size));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc
index 485c38085e..3b1f70cdb4 100644
--- a/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc
@@ -179,7 +179,6 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
   // Configure NetEq and the NetEqTest object.
   NetEqTest::Callbacks callbacks;
   NetEq::Config config;
-  config.enable_post_decode_vad = true;
   config.enable_fast_accelerate = true;
   auto codecs = NetEqTest::StandardDecoderMap();
   // rate_types contains the payload types that will be used for encoding.
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc
index ddf2ca9d3d..97b0ce2c03 100644
--- a/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_h264_fuzzer.cc
@@ -1,75 +1,75 @@
-/*

- *  Copyright (c) 2024 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 <stddef.h>

-#include <stdint.h>

-

-#include "api/video/video_frame_type.h"

-#include "modules/rtp_rtcp/source/rtp_format.h"

-#include "modules/rtp_rtcp/source/rtp_format_h264.h"

-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"

-#include "rtc_base/checks.h"

-#include "test/fuzzers/fuzz_data_helper.h"

-

-namespace webrtc {

-void FuzzOneInput(const uint8_t* data, size_t size) {

-  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));

-

-  RtpPacketizer::PayloadSizeLimits limits;

-  limits.max_payload_len = 1200;

-  // Read uint8_t to be sure reduction_lens are much smaller than

-  // max_payload_len and thus limits structure is valid.

-  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.single_packet_reduction_len =

-      fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  const H264PacketizationMode kPacketizationModes[] = {

-      H264PacketizationMode::NonInterleaved,

-      H264PacketizationMode::SingleNalUnit};

-

-  H264PacketizationMode packetization_mode =

-      fuzz_input.SelectOneOf(kPacketizationModes);

-

-  // Main function under test: RtpPacketizerH264's constructor.

-  RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),

-                               limits, packetization_mode);

-

-  size_t num_packets = packetizer.NumPackets();

-  if (num_packets == 0) {

-    return;

-  }

-  // When packetization was successful, validate NextPacket function too.

-  // While at it, check that packets respect the payload size limits.

-  RtpPacketToSend rtp_packet(nullptr);

-  // Single packet.

-  if (num_packets == 1) {

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-    RTC_CHECK_LE(rtp_packet.payload_size(),

-                 limits.max_payload_len - limits.single_packet_reduction_len);

-    return;

-  }

-  // First packet.

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.first_packet_reduction_len);

-  // Middle packets.

-  for (size_t i = 1; i < num_packets - 1; ++i) {

-    rtp_packet.Clear();

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet))

-        << "Failed to get packet#" << i;

-    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)

-        << "Packet #" << i << " exceeds it's limit";

-  }

-  // Last packet.

-  rtp_packet.Clear();

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.last_packet_reduction_len);

-}

-}  // namespace webrtc

+/*
+ *  Copyright (c) 2024 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 <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_h264.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Read uint8_t to be sure reduction_lens are much smaller than
+  // max_payload_len and thus limits structure is valid.
+  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.single_packet_reduction_len =
+      fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  const H264PacketizationMode kPacketizationModes[] = {
+      H264PacketizationMode::NonInterleaved,
+      H264PacketizationMode::SingleNalUnit};
+
+  H264PacketizationMode packetization_mode =
+      fuzz_input.SelectOneOf(kPacketizationModes);
+
+  // Main function under test: RtpPacketizerH264's constructor.
+  RtpPacketizerH264 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                               limits, packetization_mode);
+
+  size_t num_packets = packetizer.NumPackets();
+  if (num_packets == 0) {
+    return;
+  }
+  // When packetization was successful, validate NextPacket function too.
+  // While at it, check that packets respect the payload size limits.
+  RtpPacketToSend rtp_packet(nullptr);
+  // Single packet.
+  if (num_packets == 1) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+    RTC_CHECK_LE(rtp_packet.payload_size(),
+                 limits.max_payload_len - limits.single_packet_reduction_len);
+    return;
+  }
+  // First packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.first_packet_reduction_len);
+  // Middle packets.
+  for (size_t i = 1; i < num_packets - 1; ++i) {
+    rtp_packet.Clear();
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+        << "Failed to get packet#" << i;
+    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+        << "Packet #" << i << " exceeds it's limit";
+  }
+  // Last packet.
+  rtp_packet.Clear();
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.last_packet_reduction_len);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc
index c3c055de0f..93706e9253 100644
--- a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc
@@ -1,73 +1,73 @@
-/*

- *  Copyright (c) 2024 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 <stddef.h>

-#include <stdint.h>

-

-#include "api/video/video_frame_type.h"

-#include "modules/rtp_rtcp/source/rtp_format.h"

-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"

-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"

-#include "rtc_base/checks.h"

-#include "test/fuzzers/fuzz_data_helper.h"

-

-namespace webrtc {

-void FuzzOneInput(const uint8_t* data, size_t size) {

-  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));

-

-  RtpPacketizer::PayloadSizeLimits limits;

-  limits.max_payload_len = 1200;

-  // Read uint8_t to be sure reduction_lens are much smaller than

-  // max_payload_len and thus limits structure is valid.

-  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.single_packet_reduction_len =

-      fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-

-  RTPVideoHeaderVP8 hdr_info;

-  hdr_info.InitRTPVideoHeaderVP8();

-  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);

-  hdr_info.pictureId =

-      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;

-

-  // Main function under test: RtpPacketizerVp8's constructor.

-  RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),

-                              limits, hdr_info);

-

-  size_t num_packets = packetizer.NumPackets();

-  if (num_packets == 0) {

-    return;

-  }

-  // When packetization was successful, validate NextPacket function too.

-  // While at it, check that packets respect the payload size limits.

-  RtpPacketToSend rtp_packet(nullptr);

-  // Single packet.

-  if (num_packets == 1) {

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-    RTC_CHECK_LE(rtp_packet.payload_size(),

-                 limits.max_payload_len - limits.single_packet_reduction_len);

-    return;

-  }

-  // First packet.

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.first_packet_reduction_len);

-  // Middle packets.

-  for (size_t i = 1; i < num_packets - 1; ++i) {

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet))

-        << "Failed to get packet#" << i;

-    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)

-        << "Packet #" << i << " exceeds it's limit";

-  }

-  // Last packet.

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.last_packet_reduction_len);

-}

-}  // namespace webrtc

+/*
+ *  Copyright (c) 2024 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 <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Read uint8_t to be sure reduction_lens are much smaller than
+  // max_payload_len and thus limits structure is valid.
+  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.single_packet_reduction_len =
+      fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  hdr_info.pictureId =
+      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
+
+  // Main function under test: RtpPacketizerVp8's constructor.
+  RtpPacketizerVp8 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, hdr_info);
+
+  size_t num_packets = packetizer.NumPackets();
+  if (num_packets == 0) {
+    return;
+  }
+  // When packetization was successful, validate NextPacket function too.
+  // While at it, check that packets respect the payload size limits.
+  RtpPacketToSend rtp_packet(nullptr);
+  // Single packet.
+  if (num_packets == 1) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+    RTC_CHECK_LE(rtp_packet.payload_size(),
+                 limits.max_payload_len - limits.single_packet_reduction_len);
+    return;
+  }
+  // First packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.first_packet_reduction_len);
+  // Middle packets.
+  for (size_t i = 1; i < num_packets - 1; ++i) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+        << "Failed to get packet#" << i;
+    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+        << "Packet #" << i << " exceeds it's limit";
+  }
+  // Last packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.last_packet_reduction_len);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
index 3b5e67f697..d95114eaef 100644
--- a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
@@ -1,73 +1,73 @@
-/*

- *  Copyright (c) 2024 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 <stddef.h>

-#include <stdint.h>

-

-#include "api/video/video_frame_type.h"

-#include "modules/rtp_rtcp/source/rtp_format.h"

-#include "modules/rtp_rtcp/source/rtp_format_vp9.h"

-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"

-#include "rtc_base/checks.h"

-#include "test/fuzzers/fuzz_data_helper.h"

-

-namespace webrtc {

-void FuzzOneInput(const uint8_t* data, size_t size) {

-  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));

-

-  RtpPacketizer::PayloadSizeLimits limits;

-  limits.max_payload_len = 1200;

-  // Read uint8_t to be sure reduction_lens are much smaller than

-  // max_payload_len and thus limits structure is valid.

-  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-  limits.single_packet_reduction_len =

-      fuzz_input.ReadOrDefaultValue<uint8_t>(0);

-

-  RTPVideoHeaderVP9 hdr_info;

-  hdr_info.InitRTPVideoHeaderVP9();

-  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);

-  hdr_info.picture_id =

-      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;

-

-  // Main function under test: RtpPacketizerVp9's constructor.

-  RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),

-                              limits, hdr_info);

-

-  size_t num_packets = packetizer.NumPackets();

-  if (num_packets == 0) {

-    return;

-  }

-  // When packetization was successful, validate NextPacket function too.

-  // While at it, check that packets respect the payload size limits.

-  RtpPacketToSend rtp_packet(nullptr);

-  // Single packet.

-  if (num_packets == 1) {

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-    RTC_CHECK_LE(rtp_packet.payload_size(),

-                 limits.max_payload_len - limits.single_packet_reduction_len);

-    return;

-  }

-  // First packet.

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.first_packet_reduction_len);

-  // Middle packets.

-  for (size_t i = 1; i < num_packets - 1; ++i) {

-    RTC_CHECK(packetizer.NextPacket(&rtp_packet))

-        << "Failed to get packet#" << i;

-    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)

-        << "Packet #" << i << " exceeds it's limit";

-  }

-  // Last packet.

-  RTC_CHECK(packetizer.NextPacket(&rtp_packet));

-  RTC_CHECK_LE(rtp_packet.payload_size(),

-               limits.max_payload_len - limits.last_packet_reduction_len);

-}

-}  // namespace webrtc

+/*
+ *  Copyright (c) 2024 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 <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Read uint8_t to be sure reduction_lens are much smaller than
+  // max_payload_len and thus limits structure is valid.
+  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.single_packet_reduction_len =
+      fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+
+  RTPVideoHeaderVP9 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP9();
+  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  hdr_info.picture_id =
+      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
+
+  // Main function under test: RtpPacketizerVp9's constructor.
+  RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, hdr_info);
+
+  size_t num_packets = packetizer.NumPackets();
+  if (num_packets == 0) {
+    return;
+  }
+  // When packetization was successful, validate NextPacket function too.
+  // While at it, check that packets respect the payload size limits.
+  RtpPacketToSend rtp_packet(nullptr);
+  // Single packet.
+  if (num_packets == 1) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+    RTC_CHECK_LE(rtp_packet.payload_size(),
+                 limits.max_payload_len - limits.single_packet_reduction_len);
+    return;
+  }
+  // First packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.first_packet_reduction_len);
+  // Middle packets.
+  for (size_t i = 1; i < num_packets - 1; ++i) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+        << "Failed to get packet#" << i;
+    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+        << "Packet #" << i << " exceeds it's limit";
+  }
+  // Last packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.last_packet_reduction_len);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mock_audio_encoder.h b/third_party/libwebrtc/test/mock_audio_encoder.h
index 1f4510e885..30518e8f49 100644
--- a/third_party/libwebrtc/test/mock_audio_encoder.h
+++ b/third_party/libwebrtc/test/mock_audio_encoder.h
@@ -33,6 +33,10 @@ class MockAudioEncoder : public AudioEncoder {
               GetFrameLengthRange,
               (),
               (const, override));
+  MOCK_METHOD((absl::optional<std::pair<DataRate, DataRate>>),
+              GetBitrateRange,
+              (),
+              (const, override));
 
   MOCK_METHOD(void, Reset, (), (override));
   MOCK_METHOD(bool, SetFec, (bool enable), (override));
diff --git a/third_party/libwebrtc/test/mock_transformable_frame.h b/third_party/libwebrtc/test/mock_transformable_frame.h
deleted file mode 100644
index 26eb6b7030..0000000000
--- a/third_party/libwebrtc/test/mock_transformable_frame.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2020 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 TEST_MOCK_TRANSFORMABLE_FRAME_H_
-#define TEST_MOCK_TRANSFORMABLE_FRAME_H_
-
-#include "api/frame_transformer_interface.h"
-#include "test/gmock.h"
-
-namespace webrtc {
-
-class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
- public:
-  MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const, override));
-  MOCK_METHOD(rtc::ArrayView<const uint32_t>,
-              GetContributingSources,
-              (),
-              (const, override));
-  MOCK_METHOD(absl::optional<uint64_t>,
-              AbsoluteCaptureTimestamp,
-              (),
-              (const, override));
-  MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t>), (override));
-  MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
-  MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
-  MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
-  MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override));
-  MOCK_METHOD(Direction, GetDirection, (), (const, override));
-  MOCK_METHOD(std::string, GetMimeType, (), (const, override));
-};
-
-}  // namespace webrtc
-
-#endif  // TEST_MOCK_TRANSFORMABLE_FRAME_H_
diff --git a/third_party/libwebrtc/test/network/BUILD.gn b/third_party/libwebrtc/test/network/BUILD.gn
index 6df563d31d..e9bd263ed9 100644
--- a/third_party/libwebrtc/test/network/BUILD.gn
+++ b/third_party/libwebrtc/test/network/BUILD.gn
@@ -56,6 +56,7 @@ rtc_library("emulated_network") {
     "../../api/units:time_delta",
     "../../api/units:timestamp",
     "../../call:simulated_network",
+    "../../p2p:basic_packet_socket_factory",
     "../../p2p:p2p_server_utils",
     "../../p2p:rtc_p2p",
     "../../rtc_base:async_packet_socket",
@@ -128,6 +129,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../../call:simulated_network",
       "../../media:rtc_audio_video",
       "../../modules/audio_device:test_audio_device_module",
+      "../../p2p:basic_packet_socket_factory",
+      "../../p2p:basic_port_allocator",
       "../../p2p:rtc_p2p",
       "../../pc:pc_test_utils",
       "../../pc:peerconnection_wrapper",
diff --git a/third_party/libwebrtc/test/pc/e2e/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
index 0eb7aa2c68..22c9ee48d2 100644
--- a/third_party/libwebrtc/test/pc/e2e/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@@ -109,6 +109,7 @@ if (!build_with_chromium) {
         "../../../api/video_codecs:builtin_video_encoder_factory",
         "../../../modules/audio_device:test_audio_device_module",
         "../../../modules/audio_processing/aec_dump",
+        "../../../p2p:basic_port_allocator",
         "../../../p2p:rtc_p2p",
         "../../../rtc_base:threading",
         "analyzer/video:quality_analyzing_video_encoder",
@@ -576,6 +577,7 @@ if (!build_with_chromium) {
       "../../../media:media_constants",
       "../../../media:rid_description",
       "../../../media:rtc_media_base",
+      "../../../p2p:p2p_constants",
       "../../../p2p:rtc_p2p",
       "../../../pc:sdp_utils",
       "../../../pc:session_description",
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
index 17876e54be..6adfc50049 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
@@ -130,6 +130,7 @@ rtc_library("quality_analyzing_video_decoder") {
     ":encoded_image_data_injector_api",
     ":simulcast_dummy_buffer_helper",
     "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/environment",
     "../../../../../api/video:encoded_image",
     "../../../../../api/video:video_frame",
     "../../../../../api/video_codecs:video_codecs_api",
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
index e17b5d5d83..3cd179370f 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
@@ -259,10 +259,10 @@ QualityAnalyzingVideoDecoderFactory::GetSupportedFormats() const {
   return delegate_->GetSupportedFormats();
 }
 
-std::unique_ptr<VideoDecoder>
-QualityAnalyzingVideoDecoderFactory::CreateVideoDecoder(
+std::unique_ptr<VideoDecoder> QualityAnalyzingVideoDecoderFactory::Create(
+    const Environment& env,
     const SdpVideoFormat& format) {
-  std::unique_ptr<VideoDecoder> decoder = delegate_->CreateVideoDecoder(format);
+  std::unique_ptr<VideoDecoder> decoder = delegate_->Create(env, format);
   return std::make_unique<QualityAnalyzingVideoDecoder>(
       peer_name_, std::move(decoder), extractor_, analyzer_);
 }
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
index 2f0c2b9d5d..daa919d7e4 100644
--- a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
@@ -18,6 +18,7 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/environment/environment.h"
 #include "api/test/video_quality_analyzer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
@@ -136,8 +137,8 @@ class QualityAnalyzingVideoDecoderFactory : public VideoDecoderFactory {
 
   // Methods of VideoDecoderFactory interface.
   std::vector<SdpVideoFormat> GetSupportedFormats() const override;
-  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
-      const SdpVideoFormat& format) override;
+  std::unique_ptr<VideoDecoder> Create(const Environment& env,
+                                       const SdpVideoFormat& format) override;
 
  private:
   const std::string peer_name_;
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
index 257fecf309..3c4f6cabe1 100644
--- a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
@@ -27,11 +27,6 @@ using ::webrtc::test::Unit;
 
 constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
 
-// Field trial which controls whether to report standard-compliant bytes
-// sent/received per stream.  If enabled, padding and headers are not included
-// in bytes sent or received.
-constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
-
 }  // namespace
 
 NetworkQualityMetricsReporter::NetworkQualityMetricsReporter(
@@ -107,11 +102,6 @@ void NetworkQualityMetricsReporter::StopAndReportResults() {
   ReportStats(alice_network_label_, alice_stats, alice_packets_loss);
   ReportStats(bob_network_label_, bob_stats, bob_packets_loss);
 
-  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
-    RTC_LOG(LS_ERROR)
-        << "Non-standard GetStats; \"payload\" counts include RTP headers";
-  }
-
   MutexLock lock(&lock_);
   for (const auto& pair : pc_stats_) {
     ReportPCStats(pair.first, pair.second);
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
index 1348a58943..fd523cc48d 100644
--- a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
@@ -48,8 +48,6 @@ class NetworkQualityMetricsReporter
 
  private:
   struct PCStats {
-    // TODO(nisse): Separate audio and video counters. Depends on standard stat
-    // counters, enabled by field trial "WebRTC-UseStandardBytesStats".
     DataSize payload_received = DataSize::Zero();
     DataSize payload_sent = DataSize::Zero();
   };
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
index 90f201facd..3a6b808167 100644
--- a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
@@ -73,8 +73,6 @@ constexpr TimeDelta kQuickTestModeRunDuration = TimeDelta::Millis(100);
 // Field trials to enable Flex FEC advertising and receiving.
 constexpr char kFlexFecEnabledFieldTrials[] =
     "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
-constexpr char kUseStandardsBytesStats[] =
-    "WebRTC-UseStandardBytesStats/Enabled/";
 
 class FixturePeerConnectionObserver : public MockPeerConnectionObserver {
  public:
@@ -439,8 +437,7 @@ void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
 
 std::string PeerConnectionE2EQualityTest::GetFieldTrials(
     const RunParams& run_params) {
-  std::vector<absl::string_view> default_field_trials = {
-      kUseStandardsBytesStats};
+  std::vector<absl::string_view> default_field_trials = {};
   if (run_params.enable_flex_fec_support) {
     default_field_trials.push_back(kFlexFecEnabledFieldTrials);
   }
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
index b965a7acd8..706224ce08 100644
--- a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -51,11 +51,6 @@ using NetworkLayerStats =
 
 constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
 
-// Field trial which controls whether to report standard-compliant bytes
-// sent/received per stream.  If enabled, padding and headers are not included
-// in bytes sent or received.
-constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
-
 EmulatedNetworkStats PopulateStats(std::vector<EmulatedEndpoint*> endpoints,
                                    NetworkEmulationManager* network_emulation) {
   rtc::Event stats_loaded;
@@ -325,11 +320,6 @@ void StatsBasedNetworkQualityMetricsReporter::OnStatsReports(
 void StatsBasedNetworkQualityMetricsReporter::StopAndReportResults() {
   Timestamp end_time = clock_->CurrentTime();
 
-  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
-    RTC_LOG(LS_ERROR)
-        << "Non-standard GetStats; \"payload\" counts include RTP headers";
-  }
-
   std::map<std::string, NetworkLayerStats> stats = collector_.GetStats();
   for (const auto& entry : stats) {
     LogNetworkLayerStats(entry.first, entry.second);
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
index 60daf40c8c..ba6bf04e18 100644
--- a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
@@ -70,9 +70,6 @@ class StatsBasedNetworkQualityMetricsReporter
 
  private:
   struct PCStats {
-    // TODO(bugs.webrtc.org/10525): Separate audio and video counters. Depends
-    // on standard stat counters, enabled by field trial
-    // "WebRTC-UseStandardBytesStats".
     DataSize payload_received = DataSize::Zero();
     DataSize payload_sent = DataSize::Zero();
 
diff --git a/third_party/libwebrtc/test/peer_scenario/BUILD.gn b/third_party/libwebrtc/test/peer_scenario/BUILD.gn
index 18f81a56e6..e1d164a47d 100644
--- a/third_party/libwebrtc/test/peer_scenario/BUILD.gn
+++ b/third_party/libwebrtc/test/peer_scenario/BUILD.gn
@@ -53,7 +53,9 @@ if (rtc_include_tests) {
       "../../media:rtp_utils",
       "../../modules/audio_device:test_audio_device_module",
       "../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../p2p:basic_port_allocator",
       "../../p2p:rtc_p2p",
+      "../../p2p:transport_description",
       "../../pc:channel",
       "../../pc:jsep_transport_controller",
       "../../pc:pc_test_utils",
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
index 1397b32fe3..3ba4fdb677 100644
--- a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
@@ -370,10 +370,13 @@ void PeerScenarioClient::CreateAndSetSdp(
 
 void PeerScenarioClient::SetSdpOfferAndGetAnswer(
     std::string remote_offer,
+    std::function<void()> remote_description_set,
     std::function<void(std::string)> answer_handler) {
   if (!signaling_thread_->IsCurrent()) {
-    signaling_thread_->PostTask(
-        [=] { SetSdpOfferAndGetAnswer(remote_offer, answer_handler); });
+    signaling_thread_->PostTask([=] {
+      SetSdpOfferAndGetAnswer(remote_offer, remote_description_set,
+                              answer_handler);
+    });
     return;
   }
   RTC_DCHECK_RUN_ON(signaling_thread_);
@@ -381,6 +384,11 @@ void PeerScenarioClient::SetSdpOfferAndGetAnswer(
       CreateSessionDescription(SdpType::kOffer, remote_offer),
       rtc::make_ref_counted<LambdaSetRemoteDescriptionObserver>([=](RTCError) {
         RTC_DCHECK_RUN_ON(signaling_thread_);
+        if (remote_description_set) {
+          // Allow the caller to modify transceivers
+          // before creating the answer.
+          remote_description_set();
+        }
         peer_connection_->CreateAnswer(
             rtc::make_ref_counted<LambdaCreateSessionDescriptionObserver>(
                 [=](std::unique_ptr<SessionDescriptionInterface> answer) {
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h
index e863757759..cb025e9879 100644
--- a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h
@@ -147,6 +147,7 @@ class PeerScenarioClient {
       std::function<void(SessionDescriptionInterface*)> munge_offer,
       std::function<void(std::string)> offer_handler);
   void SetSdpOfferAndGetAnswer(std::string remote_offer,
+                               std::function<void()> remote_description_set,
                                std::function<void(std::string)> answer_handler);
   void SetSdpAnswer(
       std::string remote_answer,
diff --git a/third_party/libwebrtc/test/peer_scenario/signaling_route.cc b/third_party/libwebrtc/test/peer_scenario/signaling_route.cc
index eeec7c8657..8688c1abd8 100644
--- a/third_party/libwebrtc/test/peer_scenario/signaling_route.cc
+++ b/third_party/libwebrtc/test/peer_scenario/signaling_route.cc
@@ -59,6 +59,7 @@ void StartSdpNegotiation(
     CrossTrafficRoute* ret_route,
     std::function<void(SessionDescriptionInterface* offer)> munge_offer,
     std::function<void(SessionDescriptionInterface*)> modify_offer,
+    std::function<void()> callee_remote_description_set,
     std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
   caller->CreateAndSetSdp(munge_offer, [=](std::string sdp_offer) {
     if (modify_offer) {
@@ -67,11 +68,14 @@ void StartSdpNegotiation(
       RTC_CHECK(offer->ToString(&sdp_offer));
     }
     send_route->NetworkDelayedAction(kSdpPacketSize, [=] {
-      callee->SetSdpOfferAndGetAnswer(sdp_offer, [=](std::string answer) {
-        ret_route->NetworkDelayedAction(kSdpPacketSize, [=] {
-          caller->SetSdpAnswer(std::move(answer), std::move(exchange_finished));
-        });
-      });
+      callee->SetSdpOfferAndGetAnswer(
+          sdp_offer, std::move(callee_remote_description_set),
+          [=](std::string answer) {
+            ret_route->NetworkDelayedAction(kSdpPacketSize, [=] {
+              caller->SetSdpAnswer(std::move(answer),
+                                   std::move(exchange_finished));
+            });
+          });
     });
   });
 }
@@ -92,22 +96,39 @@ void SignalingRoute::StartIceSignaling() {
 }
 
 void SignalingRoute::NegotiateSdp(
+    std::function<void(SessionDescriptionInterface* offer)> munge_offer,
+    std::function<void(SessionDescriptionInterface* offer)> modify_offer,
+    std::function<void()> callee_remote_description_set,
+    std::function<void(const SessionDescriptionInterface& answer)>
+        exchange_finished) {
+  StartSdpNegotiation(caller_, callee_, send_route_, ret_route_, munge_offer,
+                      modify_offer, callee_remote_description_set,
+                      exchange_finished);
+}
+
+void SignalingRoute::NegotiateSdp(
     std::function<void(SessionDescriptionInterface*)> munge_offer,
     std::function<void(SessionDescriptionInterface*)> modify_offer,
     std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
-  StartSdpNegotiation(caller_, callee_, send_route_, ret_route_, munge_offer,
-                      modify_offer, exchange_finished);
+  NegotiateSdp(munge_offer, modify_offer, {}, exchange_finished);
 }
 
 void SignalingRoute::NegotiateSdp(
     std::function<void(SessionDescriptionInterface*)> modify_offer,
     std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
-  NegotiateSdp({}, modify_offer, exchange_finished);
+  NegotiateSdp({}, modify_offer, {}, exchange_finished);
+}
+
+void SignalingRoute::NegotiateSdp(
+    std::function<void()> remote_description_set,
+    std::function<void(const SessionDescriptionInterface& answer)>
+        exchange_finished) {
+  NegotiateSdp({}, {}, remote_description_set, exchange_finished);
 }
 
 void SignalingRoute::NegotiateSdp(
     std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
-  NegotiateSdp({}, {}, exchange_finished);
+  NegotiateSdp({}, {}, {}, exchange_finished);
 }
 
 }  // namespace test
diff --git a/third_party/libwebrtc/test/peer_scenario/signaling_route.h b/third_party/libwebrtc/test/peer_scenario/signaling_route.h
index a95ae5c9f7..9b317d2552 100644
--- a/third_party/libwebrtc/test/peer_scenario/signaling_route.h
+++ b/third_party/libwebrtc/test/peer_scenario/signaling_route.h
@@ -35,12 +35,21 @@ class SignalingRoute {
   // The `munge_offer` callback is used to modify an offer between its creation
   // and set local description. This behavior is forbidden according to the spec
   // but available here in order to allow test coverage on corner cases.
-  // The `exchange_finished` callback is called with the answer produced after
-  // SDP negotations has completed.
+  // `callee_remote_description_set` is invoked when callee has applied the
+  // offer but not yet created an answer. The purpose is to allow tests to
+  // modify transceivers created from the offer.  The `exchange_finished`
+  // callback is called with the answer produced after SDP negotations has
+  // completed.
   // TODO(srte): Handle lossy links.
   void NegotiateSdp(
       std::function<void(SessionDescriptionInterface* offer)> munge_offer,
       std::function<void(SessionDescriptionInterface* offer)> modify_offer,
+      std::function<void()> callee_remote_description_set,
+      std::function<void(const SessionDescriptionInterface& answer)>
+          exchange_finished);
+  void NegotiateSdp(
+      std::function<void(SessionDescriptionInterface* offer)> munge_offer,
+      std::function<void(SessionDescriptionInterface* offer)> modify_offer,
       std::function<void(const SessionDescriptionInterface& answer)>
           exchange_finished);
   void NegotiateSdp(
@@ -48,6 +57,10 @@ class SignalingRoute {
       std::function<void(const SessionDescriptionInterface& answer)>
           exchange_finished);
   void NegotiateSdp(
+      std::function<void()> remote_description_set,
+      std::function<void(const SessionDescriptionInterface& answer)>
+          exchange_finished);
+  void NegotiateSdp(
       std::function<void(const SessionDescriptionInterface& answer)>
           exchange_finished);
   SignalingRoute reverse() {
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/bwe_ramp_up_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/bwe_ramp_up_test.cc
index a7a17bbfd1..f8eaa47858 100644
--- a/third_party/libwebrtc/test/peer_scenario/tests/bwe_ramp_up_test.cc
+++ b/third_party/libwebrtc/test/peer_scenario/tests/bwe_ramp_up_test.cc
@@ -25,6 +25,9 @@ namespace webrtc {
 namespace test {
 
 using ::testing::SizeIs;
+using ::testing::Test;
+using ::testing::ValuesIn;
+using ::testing::WithParamInterface;
 
 rtc::scoped_refptr<const RTCStatsReport> GetStatsAndProcess(
     PeerScenario& s,
@@ -124,5 +127,152 @@ TEST(BweRampupTest, RampUpWithUndemuxableRtpPackets) {
   // ensure BWE has increased beyond noise levels.
   EXPECT_GT(final_bwe, initial_bwe + DataRate::KilobitsPerSec(345));
 }
+
+struct InitialProbeTestParams {
+  DataRate network_capacity;
+  DataRate expected_bwe_min;
+};
+class BweRampupWithInitialProbeTest
+    : public Test,
+      public WithParamInterface<InitialProbeTestParams> {};
+
+INSTANTIATE_TEST_SUITE_P(
+    BweRampupWithInitialProbeTest,
+    BweRampupWithInitialProbeTest,
+    ValuesIn<InitialProbeTestParams>(
+        {{
+             .network_capacity = DataRate::KilobitsPerSec(3000),
+             .expected_bwe_min = DataRate::KilobitsPerSec(2500),
+         },
+         {
+             .network_capacity = webrtc::DataRate::KilobitsPerSec(500),
+             .expected_bwe_min = webrtc::DataRate::KilobitsPerSec(400),
+         }}));
+
+// Test that caller and callee BWE rampup even if no media packets are sent.
+// - BandWidthEstimationSettings.allow_probe_without_media must be set.
+// - A Video RtpTransceiver with RTX support needs to be negotiated.
+TEST_P(BweRampupWithInitialProbeTest, BweRampUpBothDirectionsWithoutMedia) {
+  PeerScenario s(*::testing::UnitTest::GetInstance()->current_test_info());
+  InitialProbeTestParams test_params = GetParam();
+
+  PeerScenarioClient* caller = s.CreateClient({});
+  PeerScenarioClient* callee = s.CreateClient({});
+
+  auto video_result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
+  ASSERT_EQ(video_result.error().type(), RTCErrorType::NONE);
+
+  caller->pc()->ReconfigureBandwidthEstimation(
+      {.allow_probe_without_media = true});
+  callee->pc()->ReconfigureBandwidthEstimation(
+      {.allow_probe_without_media = true});
+
+  auto node_builder =
+      s.net()->NodeBuilder().capacity_kbps(test_params.network_capacity.kbps());
+  auto caller_node = node_builder.Build().node;
+  auto callee_node = node_builder.Build().node;
+  s.net()->CreateRoute(caller->endpoint(), {caller_node}, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), {callee_node}, caller->endpoint());
+
+  auto signaling =
+      s.ConnectSignaling(caller, callee, {caller_node}, {callee_node});
+  signaling.StartIceSignaling();
+
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp(
+      [&]() {
+        // When remote description has been set, a transceiver is created.
+        // Set the diretion to sendrecv so that it can be used for BWE probing
+        // from callee -> caller.
+        ASSERT_THAT(callee->pc()->GetTransceivers(), SizeIs(1));
+        ASSERT_TRUE(
+            callee->pc()
+                ->GetTransceivers()[0]
+                ->SetDirectionWithError(RtpTransceiverDirection::kSendRecv)
+                .ok());
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        offer_exchange_done = true;
+      });
+  // Wait for SDP negotiation.
+  s.WaitAndProcess(&offer_exchange_done);
+
+  // Test that 1s after offer/answer exchange finish, we have a BWE estimate,
+  // even though no video frames have been sent.
+  s.ProcessMessages(TimeDelta::Seconds(1));
+
+  auto callee_inbound_stats =
+      GetStatsAndProcess(s, callee)->GetStatsOfType<RTCInboundRtpStreamStats>();
+  ASSERT_THAT(callee_inbound_stats, SizeIs(1));
+  ASSERT_EQ(*callee_inbound_stats[0]->frames_received, 0u);
+  auto caller_inbound_stats =
+      GetStatsAndProcess(s, caller)->GetStatsOfType<RTCInboundRtpStreamStats>();
+  ASSERT_THAT(caller_inbound_stats, SizeIs(1));
+  ASSERT_EQ(*caller_inbound_stats[0]->frames_received, 0u);
+
+  DataRate caller_bwe = GetAvailableSendBitrate(GetStatsAndProcess(s, caller));
+  EXPECT_GT(caller_bwe.kbps(), test_params.expected_bwe_min.kbps());
+  EXPECT_LE(caller_bwe.kbps(), test_params.network_capacity.kbps());
+  DataRate callee_bwe = GetAvailableSendBitrate(GetStatsAndProcess(s, callee));
+  EXPECT_GT(callee_bwe.kbps(), test_params.expected_bwe_min.kbps());
+  EXPECT_LE(callee_bwe.kbps(), test_params.network_capacity.kbps());
+}
+
+// Test that we can reconfigure bandwidth estimation and send new BWE probes.
+// In this test, camera is stopped, and some times later, the app want to get a
+// new BWE estimate.
+TEST(BweRampupTest, CanReconfigureBweAfterStopingVideo) {
+  PeerScenario s(*::testing::UnitTest::GetInstance()->current_test_info());
+  PeerScenarioClient* caller = s.CreateClient({});
+  PeerScenarioClient* callee = s.CreateClient({});
+
+  auto node_builder = s.net()->NodeBuilder().capacity_kbps(1000);
+  auto caller_node = node_builder.Build().node;
+  auto callee_node = node_builder.Build().node;
+  s.net()->CreateRoute(caller->endpoint(), {caller_node}, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), {callee_node}, caller->endpoint());
+
+  PeerScenarioClient::VideoSendTrack track = caller->CreateVideo("VIDEO", {});
+
+  auto signaling =
+      s.ConnectSignaling(caller, callee, {caller_node}, {callee_node});
+
+  signaling.StartIceSignaling();
+
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp([&](const SessionDescriptionInterface& answer) {
+    offer_exchange_done = true;
+  });
+  // Wait for SDP negotiation.
+  s.WaitAndProcess(&offer_exchange_done);
+
+  // Send a TCP messages to the receiver using the same downlink node.
+  // This is done just to force a lower BWE than the link capacity.
+  webrtc::TcpMessageRoute* tcp_route = s.net()->CreateTcpRoute(
+      s.net()->CreateRoute({caller_node}), s.net()->CreateRoute({callee_node}));
+  DataRate bwe_before_restart = DataRate::Zero();
+
+  std::atomic<bool> message_delivered(false);
+  tcp_route->SendMessage(
+      /*size=*/5'00'000,
+      /*on_received=*/[&]() { message_delivered = true; });
+  s.WaitAndProcess(&message_delivered);
+  bwe_before_restart = GetAvailableSendBitrate(GetStatsAndProcess(s, caller));
+
+  // Camera is stopped.
+  track.capturer->Stop();
+  s.ProcessMessages(TimeDelta::Seconds(2));
+
+  // Some time later, the app is interested in restarting BWE since we may want
+  // to resume video eventually.
+  caller->pc()->ReconfigureBandwidthEstimation(
+      {.allow_probe_without_media = true});
+  s.ProcessMessages(TimeDelta::Seconds(1));
+  DataRate bwe_after_restart =
+      GetAvailableSendBitrate(GetStatsAndProcess(s, caller));
+  EXPECT_GT(bwe_after_restart.kbps(), bwe_before_restart.kbps() + 300);
+  EXPECT_LT(bwe_after_restart.kbps(), 1000);
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc
index 4f478b4b2a..dced274e68 100644
--- a/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc
+++ b/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc
@@ -98,7 +98,6 @@ TEST_P(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) {
   PeerScenarioClient::Config config = PeerScenarioClient::Config();
   // Disable encryption so that we can inject a fake early media packet without
   // triggering srtp failures.
-  config.disable_encryption = true;
   auto* caller = s.CreateClient(config);
   auto* callee = s.CreateClient(config);
 
diff --git a/third_party/libwebrtc/test/scenario/video_stream.cc b/third_party/libwebrtc/test/scenario/video_stream.cc
index 654aed7c6c..c3f0da7cb7 100644
--- a/third_party/libwebrtc/test/scenario/video_stream.cc
+++ b/third_party/libwebrtc/test/scenario/video_stream.cc
@@ -491,10 +491,6 @@ void SendVideoStream::UpdateConfig(
 void SendVideoStream::UpdateActiveLayers(std::vector<bool> active_layers) {
   sender_->task_queue_.PostTask([=] {
     MutexLock lock(&mutex_);
-    if (config_.encoder.codec ==
-        VideoStreamConfig::Encoder::Codec::kVideoCodecVP8) {
-      send_stream_->StartPerRtpStream(active_layers);
-    }
     VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config_);
     RTC_CHECK_EQ(encoder_config.simulcast_layers.size(), active_layers.size());
     for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i)
diff --git a/third_party/libwebrtc/test/testsupport/file_utils.cc b/third_party/libwebrtc/test/testsupport/file_utils.cc
index 47fed9ac05..afabbaad3f 100644
--- a/third_party/libwebrtc/test/testsupport/file_utils.cc
+++ b/third_party/libwebrtc/test/testsupport/file_utils.cc
@@ -36,7 +36,7 @@
 
 #include <sys/stat.h>  // To check for directory existence.
 #ifndef S_ISDIR        // Not defined in stat.h on Windows.
-#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
+#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
 #endif
 
 #include <stdio.h>
@@ -54,6 +54,7 @@
 
 #include "absl/strings/string_view.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/helpers.h"
 #include "rtc_base/string_utils.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/testsupport/file_utils_override.h"
@@ -94,6 +95,13 @@ std::string OutputPath() {
   return webrtc::test::internal::OutputPath();
 }
 
+std::string OutputPathWithRandomDirectory() {
+  std::string path = webrtc::test::internal::OutputPath();
+  std::string rand_dir = path + rtc::CreateRandomUuid();
+
+  return CreateDir(rand_dir) ? rand_dir + std::string(kPathDelimiter) : path;
+}
+
 std::string WorkingDir() {
   return webrtc::test::internal::WorkingDir();
 }
@@ -229,7 +237,12 @@ std::string ResourcePath(absl::string_view name, absl::string_view extension) {
 std::string JoinFilename(absl::string_view dir, absl::string_view name) {
   RTC_CHECK(!dir.empty()) << "Special cases not implemented.";
   rtc::StringBuilder os;
-  os << dir << kPathDelimiter << name;
+  os << dir;
+  // If the directory path already ends with a path delimiter don't append it
+  if (dir.back() != kPathDelimiter.back()) {
+    os << kPathDelimiter;
+  }
+  os << name;
   return os.Release();
 }
 
diff --git a/third_party/libwebrtc/test/testsupport/file_utils.h b/third_party/libwebrtc/test/testsupport/file_utils.h
index ab80ca4454..120c6cb279 100644
--- a/third_party/libwebrtc/test/testsupport/file_utils.h
+++ b/third_party/libwebrtc/test/testsupport/file_utils.h
@@ -42,6 +42,11 @@ ABSL_CONST_INIT extern const absl::string_view kPathDelimiter;
 // found, the current working directory ("./") is returned as a fallback.
 std::string OutputPath();
 
+// Same as the above but appends a randomly named folder at the end of the path
+// Primerly used to provide a solution for stress testing environments to
+// prevent colission of files and folders.
+std::string OutputPathWithRandomDirectory();
+
 // Generates an empty file with a unique name in the specified directory and
 // returns the file name and path.
 // TODO(titovartem) rename to TempFile and next method to TempFilename
diff --git a/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc b/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc
index b9de01d09d..1101a63352 100644
--- a/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc
+++ b/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc
@@ -19,6 +19,7 @@
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/helpers.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 
@@ -119,6 +120,28 @@ TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) {
   ASSERT_THAT(result, EndsWith(expected_end));
 }
 
+TEST_F(FileUtilsTest, RandomOutputPathFromUnchangedWorkingDir) {
+  rtc::SetRandomTestMode(true);
+  std::string fixed_first_uuid = "def01482-f829-429a-bfd4-841706e92cdd";
+  std::string expected_end = ExpectedRootDirByPlatform() + fixed_first_uuid +
+                             std::string(kPathDelimiter);
+  std::string result = webrtc::test::OutputPathWithRandomDirectory();
+
+  ASSERT_THAT(result, EndsWith(expected_end));
+}
+
+TEST_F(FileUtilsTest, RandomOutputPathFromRootWorkingDir) {
+  ASSERT_EQ(0, chdir(kPathDelimiter.data()));
+
+  rtc::SetRandomTestMode(true);
+  std::string fixed_first_uuid = "def01482-f829-429a-bfd4-841706e92cdd";
+  std::string expected_end = ExpectedRootDirByPlatform() + fixed_first_uuid +
+                             std::string(kPathDelimiter);
+  std::string result = webrtc::test::OutputPathWithRandomDirectory();
+
+  ASSERT_THAT(result, EndsWith(expected_end));
+}
+
 TEST_F(FileUtilsTest, TempFilename) {
   std::string temp_filename = webrtc::test::TempFilename(
       webrtc::test::OutputPath(), "TempFilenameTest");
@@ -147,7 +170,8 @@ TEST_F(FileUtilsTest, GenerateTempFilename) {
 #define MAYBE_CreateDir CreateDir
 #endif
 TEST_F(FileUtilsTest, MAYBE_CreateDir) {
-  std::string directory = "fileutils-unittest-empty-dir";
+  std::string directory =
+      test::OutputPathWithRandomDirectory() + "fileutils-unittest-empty-dir";
   // Make sure it's removed if a previous test has failed:
   remove(directory.c_str());
   ASSERT_TRUE(webrtc::test::CreateDir(directory));
@@ -231,7 +255,7 @@ TEST_F(FileUtilsTest, WriteReadDeleteFilesAndDirs) {
 
   // Create an empty temporary directory for this test.
   const std::string temp_directory =
-      OutputPath() + Path("TempFileUtilsTestReadDirectory/");
+      OutputPathWithRandomDirectory() + Path("TempFileUtilsTestReadDirectory/");
   CreateDir(temp_directory);
   EXPECT_NO_FATAL_FAILURE(CleanDir(temp_directory, &num_deleted_entries));
   EXPECT_TRUE(DirExists(temp_directory));
@@ -273,5 +297,15 @@ TEST_F(FileUtilsTest, DirNameStopsAtRoot) {
   EXPECT_EQ(Path("/"), DirName(Path("/")));
 }
 
+TEST_F(FileUtilsTest, JoinFilenameDoesNotAppendExtraPathDelimiterIfExists) {
+  EXPECT_EQ(JoinFilename(Path("/some/path/"), "file.txt"),
+            Path("/some/path/file.txt"));
+}
+
+TEST_F(FileUtilsTest, JoinFilenameAppendsPathDelimiterIfMissing) {
+  EXPECT_EQ(JoinFilename(Path("/some/path"), "file.txt"),
+            Path("/some/path/file.txt"));
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/test_artifacts.cc b/third_party/libwebrtc/test/testsupport/test_artifacts.cc
index 6f062e5fe4..b0ab046e63 100644
--- a/third_party/libwebrtc/test/testsupport/test_artifacts.cc
+++ b/third_party/libwebrtc/test/testsupport/test_artifacts.cc
@@ -20,7 +20,7 @@
 
 namespace {
 const std::string& DefaultArtifactPath() {
-  static const std::string path = webrtc::test::OutputPath();
+  static const std::string path = webrtc::test::OutputPathWithRandomDirectory();
   return path;
 }
 }  // namespace
@@ -55,8 +55,11 @@ bool WriteToTestArtifactsDir(const char* filename,
     return false;
   }
 
-  FileWrapper output = FileWrapper::OpenWriteOnly(
-      JoinFilename(absl::GetFlag(FLAGS_test_artifacts_dir), filename));
+  std::string full_path =
+      JoinFilename(absl::GetFlag(FLAGS_test_artifacts_dir), filename);
+  FileWrapper output = FileWrapper::OpenWriteOnly(full_path);
+
+  RTC_LOG(LS_INFO) << "Writing test artifacts in: " << full_path;
 
   return output.is_open() && output.Write(buffer, length);
 }
diff --git a/third_party/libwebrtc/test/video_codec_tester.cc b/third_party/libwebrtc/test/video_codec_tester.cc
index f5fdc07a6b..9aef46d44e 100644
--- a/third_party/libwebrtc/test/video_codec_tester.cc
+++ b/third_party/libwebrtc/test/video_codec_tester.cc
@@ -772,10 +772,12 @@ class VideoCodecAnalyzer : public VideoCodecTester::VideoCodecStats {
 
 class Decoder : public DecodedImageCallback {
  public:
-  Decoder(VideoDecoderFactory* decoder_factory,
+  Decoder(const Environment& env,
+          VideoDecoderFactory* decoder_factory,
           const DecoderSettings& decoder_settings,
           VideoCodecAnalyzer* analyzer)
-      : decoder_factory_(decoder_factory),
+      : env_(env),
+        decoder_factory_(decoder_factory),
         analyzer_(analyzer),
         pacer_(decoder_settings.pacing_settings) {
     RTC_CHECK(analyzer_) << "Analyzer must be provided";
@@ -792,7 +794,7 @@ class Decoder : public DecodedImageCallback {
   }
 
   void Initialize(const SdpVideoFormat& sdp_video_format) {
-    decoder_ = decoder_factory_->CreateVideoDecoder(sdp_video_format);
+    decoder_ = decoder_factory_->Create(env_, sdp_video_format);
     RTC_CHECK(decoder_) << "Could not create decoder for video format "
                         << sdp_video_format.ToString();
 
@@ -863,6 +865,7 @@ class Decoder : public DecodedImageCallback {
     return WEBRTC_VIDEO_CODEC_OK;
   }
 
+  const Environment env_;
   VideoDecoderFactory* decoder_factory_;
   std::unique_ptr<VideoDecoder> decoder_;
   VideoCodecAnalyzer* const analyzer_;
@@ -982,7 +985,8 @@ class Encoder : public EncodedImageCallback {
       // layer >X receive encoded lower layers.
       int num_spatial_layers =
           ScalabilityModeToNumSpatialLayers(last_superframe_->scalability_mode);
-      for (int sidx = *last_superframe_->encoded_frame.SpatialIndex() + 1;
+      for (int sidx =
+               last_superframe_->encoded_frame.SpatialIndex().value_or(0) + 1;
            sidx < num_spatial_layers; ++sidx) {
         last_superframe_->encoded_frame.SetSpatialIndex(sidx);
         DeliverEncodedFrame(last_superframe_->encoded_frame);
@@ -1108,8 +1112,6 @@ class Encoder : public EncodedImageCallback {
 
     int result = encoder_->InitEncode(&vc, ves);
     RTC_CHECK(result == WEBRTC_VIDEO_CODEC_OK);
-
-    SetRates(es);
   }
 
   void SetRates(const EncodingSettings& es) {
@@ -1258,6 +1260,29 @@ void ConfigureSimulcast(VideoCodec* vc) {
   }
 }
 
+void SetDefaultCodecSpecificSettings(VideoCodec* vc, int num_temporal_layers) {
+  switch (vc->codecType) {
+    case kVideoCodecVP8:
+      *(vc->VP8()) = VideoEncoder::GetDefaultVp8Settings();
+      vc->VP8()->SetNumberOfTemporalLayers(num_temporal_layers);
+      break;
+    case kVideoCodecVP9: {
+      *(vc->VP9()) = VideoEncoder::GetDefaultVp9Settings();
+      vc->VP9()->SetNumberOfTemporalLayers(num_temporal_layers);
+    } break;
+    case kVideoCodecH264: {
+      *(vc->H264()) = VideoEncoder::GetDefaultH264Settings();
+      vc->H264()->SetNumberOfTemporalLayers(num_temporal_layers);
+    } break;
+    case kVideoCodecAV1:
+    case kVideoCodecH265:
+      break;
+    case kVideoCodecGeneric:
+    case kVideoCodecMultiplex:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
 std::tuple<std::vector<DataRate>, ScalabilityMode>
 SplitBitrateAndUpdateScalabilityMode(std::string codec_type,
                                      ScalabilityMode scalability_mode,
@@ -1269,11 +1294,11 @@ SplitBitrateAndUpdateScalabilityMode(std::string codec_type,
   int num_temporal_layers =
       ScalabilityModeToNumTemporalLayers(scalability_mode);
 
-  if (bitrates_kbps.size() > 1 ||
-      (num_spatial_layers == 1 && num_temporal_layers == 1)) {
-    RTC_CHECK(bitrates_kbps.size() ==
-              static_cast<size_t>(num_spatial_layers * num_temporal_layers))
-        << "bitrates must be provided for all layers";
+  int num_bitrates = static_cast<int>(bitrates_kbps.size());
+  RTC_CHECK(num_bitrates == 1 || num_bitrates == num_spatial_layers ||
+            num_bitrates == num_spatial_layers * num_temporal_layers);
+
+  if (num_bitrates == num_spatial_layers * num_temporal_layers) {
     std::vector<DataRate> bitrates;
     for (const auto& bitrate_kbps : bitrates_kbps) {
       bitrates.push_back(DataRate::KilobitsPerSec(bitrate_kbps));
@@ -1281,59 +1306,93 @@ SplitBitrateAndUpdateScalabilityMode(std::string codec_type,
     return std::make_tuple(bitrates, scalability_mode);
   }
 
+  int total_bitrate_kbps =
+      std::accumulate(bitrates_kbps.begin(), bitrates_kbps.end(), 0);
+
   VideoCodec vc;
   vc.codecType = PayloadStringToCodecType(codec_type);
   vc.width = width;
   vc.height = height;
-  vc.startBitrate = bitrates_kbps.front();
-  vc.maxBitrate = bitrates_kbps.front();
+  vc.startBitrate = total_bitrate_kbps;
+  vc.maxBitrate = total_bitrate_kbps;
   vc.minBitrate = 0;
   vc.maxFramerate = static_cast<uint32_t>(framerate_fps);
   vc.numberOfSimulcastStreams = 0;
   vc.mode = webrtc::VideoCodecMode::kRealtimeVideo;
   vc.SetScalabilityMode(scalability_mode);
+  SetDefaultCodecSpecificSettings(&vc, num_temporal_layers);
 
-  switch (vc.codecType) {
-    case kVideoCodecVP8:
-      // TODO(webrtc:14852): Configure simulcast.
-      *(vc.VP8()) = VideoEncoder::GetDefaultVp8Settings();
-      vc.VP8()->SetNumberOfTemporalLayers(num_temporal_layers);
-      ConfigureSimulcast(&vc);
-      break;
-    case kVideoCodecVP9: {
-      *(vc.VP9()) = VideoEncoder::GetDefaultVp9Settings();
-      vc.VP9()->SetNumberOfTemporalLayers(num_temporal_layers);
-      const std::vector<SpatialLayer> spatialLayers = GetVp9SvcConfig(vc);
-      for (size_t i = 0; i < spatialLayers.size(); ++i) {
-        vc.spatialLayers[i] = spatialLayers[i];
-        vc.spatialLayers[i].active = true;
-      }
-    } break;
-    case kVideoCodecAV1: {
-      bool result =
-          SetAv1SvcConfig(vc, num_spatial_layers, num_temporal_layers);
-      RTC_CHECK(result) << "SetAv1SvcConfig failed";
-    } break;
-    case kVideoCodecH264: {
-      *(vc.H264()) = VideoEncoder::GetDefaultH264Settings();
-      vc.H264()->SetNumberOfTemporalLayers(num_temporal_layers);
-      ConfigureSimulcast(&vc);
-    } break;
-    case kVideoCodecH265:
-      break;
-    case kVideoCodecGeneric:
-    case kVideoCodecMultiplex:
-      RTC_CHECK_NOTREACHED();
-  }
+  if (num_bitrates == num_spatial_layers) {
+    switch (vc.codecType) {
+      case kVideoCodecVP8:
+      case kVideoCodecH264:
+      case kVideoCodecH265:
+        vc.numberOfSimulcastStreams = num_spatial_layers;
+        for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+          SimulcastStream* ss = &vc.simulcastStream[sidx];
+          ss->width = width >> (num_spatial_layers - sidx - 1);
+          ss->height = height >> (num_spatial_layers - sidx - 1);
+          ss->maxFramerate = vc.maxFramerate;
+          ss->numberOfTemporalLayers = num_temporal_layers;
+          ss->maxBitrate = bitrates_kbps[sidx];
+          ss->targetBitrate = bitrates_kbps[sidx];
+          ss->minBitrate = 0;
+          ss->qpMax = 0;
+          ss->active = true;
+        }
+        break;
+      case kVideoCodecVP9:
+      case kVideoCodecAV1:
+        for (int sidx = num_spatial_layers - 1; sidx >= 0; --sidx) {
+          SpatialLayer* ss = &vc.spatialLayers[sidx];
+          ss->width = width >> (num_spatial_layers - sidx - 1);
+          ss->height = height >> (num_spatial_layers - sidx - 1);
+          ss->maxFramerate = vc.maxFramerate;
+          ss->numberOfTemporalLayers = num_temporal_layers;
+          ss->maxBitrate = bitrates_kbps[sidx];
+          ss->targetBitrate = bitrates_kbps[sidx];
+          ss->minBitrate = 0;
+          ss->qpMax = 0;
+          ss->active = true;
+        }
+        break;
+      case kVideoCodecGeneric:
+      case kVideoCodecMultiplex:
+        RTC_CHECK_NOTREACHED();
+    }
+  } else {
+    switch (vc.codecType) {
+      case kVideoCodecVP8:
+      case kVideoCodecH264:
+      case kVideoCodecH265:
+        ConfigureSimulcast(&vc);
+        break;
+      case kVideoCodecVP9: {
+        const std::vector<SpatialLayer> spatialLayers = GetVp9SvcConfig(vc);
+        for (size_t i = 0; i < spatialLayers.size(); ++i) {
+          vc.spatialLayers[i] = spatialLayers[i];
+          vc.spatialLayers[i].active = true;
+        }
+      } break;
+      case kVideoCodecAV1: {
+        bool result =
+            SetAv1SvcConfig(vc, num_spatial_layers, num_temporal_layers);
+        RTC_CHECK(result) << "SetAv1SvcConfig failed";
+      } break;
+      case kVideoCodecGeneric:
+      case kVideoCodecMultiplex:
+        RTC_CHECK_NOTREACHED();
+    }
 
-  if (*vc.GetScalabilityMode() != scalability_mode) {
-    RTC_LOG(LS_WARNING) << "Scalability mode changed from "
-                        << ScalabilityModeToString(scalability_mode) << " to "
-                        << ScalabilityModeToString(*vc.GetScalabilityMode());
-    num_spatial_layers =
-        ScalabilityModeToNumSpatialLayers(*vc.GetScalabilityMode());
-    num_temporal_layers =
-        ScalabilityModeToNumTemporalLayers(*vc.GetScalabilityMode());
+    if (*vc.GetScalabilityMode() != scalability_mode) {
+      RTC_LOG(LS_WARNING) << "Scalability mode changed from "
+                          << ScalabilityModeToString(scalability_mode) << " to "
+                          << ScalabilityModeToString(*vc.GetScalabilityMode());
+      num_spatial_layers =
+          ScalabilityModeToNumSpatialLayers(*vc.GetScalabilityMode());
+      num_temporal_layers =
+          ScalabilityModeToNumTemporalLayers(*vc.GetScalabilityMode());
+    }
   }
 
   std::unique_ptr<VideoBitrateAllocator> bitrate_allocator =
@@ -1341,7 +1400,7 @@ SplitBitrateAndUpdateScalabilityMode(std::string codec_type,
           vc);
   VideoBitrateAllocation bitrate_allocation =
       bitrate_allocator->Allocate(VideoBitrateAllocationParameters(
-          1000 * bitrates_kbps.front(), framerate_fps));
+          1000 * total_bitrate_kbps, framerate_fps));
 
   std::vector<DataRate> bitrates;
   for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
@@ -1476,13 +1535,14 @@ std::map<uint32_t, EncodingSettings> VideoCodecTester::CreateEncodingSettings(
 }
 
 std::unique_ptr<VideoCodecTester::VideoCodecStats>
-VideoCodecTester::RunDecodeTest(CodedVideoSource* video_source,
+VideoCodecTester::RunDecodeTest(const Environment& env,
+                                CodedVideoSource* video_source,
                                 VideoDecoderFactory* decoder_factory,
                                 const DecoderSettings& decoder_settings,
                                 const SdpVideoFormat& sdp_video_format) {
   std::unique_ptr<VideoCodecAnalyzer> analyzer =
       std::make_unique<VideoCodecAnalyzer>(/*video_source=*/nullptr);
-  Decoder decoder(decoder_factory, decoder_settings, analyzer.get());
+  Decoder decoder(env, decoder_factory, decoder_settings, analyzer.get());
   decoder.Initialize(sdp_video_format);
 
   while (auto frame = video_source->PullFrame()) {
@@ -1522,6 +1582,7 @@ VideoCodecTester::RunEncodeTest(
 
 std::unique_ptr<VideoCodecTester::VideoCodecStats>
 VideoCodecTester::RunEncodeDecodeTest(
+    const Environment& env,
     const VideoSourceSettings& source_settings,
     VideoEncoderFactory* encoder_factory,
     VideoDecoderFactory* decoder_factory,
@@ -1539,8 +1600,8 @@ VideoCodecTester::RunEncodeDecodeTest(
       ScalabilityModeToNumSpatialLayers(frame_settings.scalability_mode);
   std::vector<std::unique_ptr<Decoder>> decoders;
   for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
-    auto decoder = std::make_unique<Decoder>(decoder_factory, decoder_settings,
-                                             analyzer.get());
+    auto decoder = std::make_unique<Decoder>(env, decoder_factory,
+                                             decoder_settings, analyzer.get());
     decoder->Initialize(frame_settings.sdp_video_format);
     decoders.push_back(std::move(decoder));
   }
diff --git a/third_party/libwebrtc/test/video_codec_tester.h b/third_party/libwebrtc/test/video_codec_tester.h
index 87cc5f76f8..00b6093dca 100644
--- a/third_party/libwebrtc/test/video_codec_tester.h
+++ b/third_party/libwebrtc/test/video_codec_tester.h
@@ -199,6 +199,7 @@ class VideoCodecTester {
 
   // Decodes video, collects and returns decode metrics.
   static std::unique_ptr<VideoCodecStats> RunDecodeTest(
+      const Environment& env,
       CodedVideoSource* video_source,
       VideoDecoderFactory* decoder_factory,
       const DecoderSettings& decoder_settings,
@@ -213,6 +214,7 @@ class VideoCodecTester {
 
   // Encodes and decodes video, collects and returns encode and decode metrics.
   static std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
+      const Environment& env,
       const VideoSourceSettings& source_settings,
       VideoEncoderFactory* encoder_factory,
       VideoDecoderFactory* decoder_factory,
diff --git a/third_party/libwebrtc/test/video_codec_tester_unittest.cc b/third_party/libwebrtc/test/video_codec_tester_unittest.cc
index df5dca90a2..fdd7b37a00 100644
--- a/third_party/libwebrtc/test/video_codec_tester_unittest.cc
+++ b/third_party/libwebrtc/test/video_codec_tester_unittest.cc
@@ -17,6 +17,8 @@
 #include <utility>
 #include <vector>
 
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/test/mock_video_decoder.h"
 #include "api/test/mock_video_decoder_factory.h"
 #include "api/test/mock_video_encoder.h"
@@ -44,13 +46,12 @@ namespace {
 using ::testing::_;
 using ::testing::ElementsAre;
 using ::testing::Field;
-using ::testing::Invoke;
-using ::testing::InvokeWithoutArgs;
 using ::testing::NiceMock;
 using ::testing::Return;
 using ::testing::SizeIs;
 using ::testing::UnorderedElementsAreArray;
 using ::testing::Values;
+using ::testing::WithoutArgs;
 
 using VideoCodecStats = VideoCodecTester::VideoCodecStats;
 using VideoSourceSettings = VideoCodecTester::VideoSourceSettings;
@@ -200,30 +201,27 @@ class VideoCodecTesterTest : public ::testing::Test {
         });
 
     NiceMock<MockVideoDecoderFactory> decoder_factory;
-    ON_CALL(decoder_factory, CreateVideoDecoder)
-        .WillByDefault([&](const SdpVideoFormat&) {
-          // Video codec tester destroyes decoder at the end of test. Test
-          // decoder collects stats which we need to access after test. To keep
-          // the decode alive we wrap it into a wrapper and pass the wrapper to
-          // the tester.
-          class DecoderWrapper : public TestVideoDecoder {
-           public:
-            explicit DecoderWrapper(TestVideoDecoder* decoder)
-                : decoder_(decoder) {}
-            int32_t Decode(const EncodedImage& encoded_frame,
-                           int64_t render_time_ms) {
-              return decoder_->Decode(encoded_frame, render_time_ms);
-            }
-            int32_t RegisterDecodeCompleteCallback(
-                DecodedImageCallback* callback) {
-              return decoder_->RegisterDecodeCompleteCallback(callback);
-            }
-            TestVideoDecoder* decoder_;
-          };
-          decoders_.push_back(std::make_unique<NiceMock<TestVideoDecoder>>());
-          return std::make_unique<NiceMock<DecoderWrapper>>(
-              decoders_.back().get());
-        });
+    ON_CALL(decoder_factory, Create).WillByDefault(WithoutArgs([&] {
+      // Video codec tester destroyes decoder at the end of test. Test
+      // decoder collects stats which we need to access after test. To keep
+      // the decode alive we wrap it into a wrapper and pass the wrapper to
+      // the tester.
+      class DecoderWrapper : public TestVideoDecoder {
+       public:
+        explicit DecoderWrapper(TestVideoDecoder* decoder)
+            : decoder_(decoder) {}
+        int32_t Decode(const EncodedImage& encoded_frame,
+                       int64_t render_time_ms) {
+          return decoder_->Decode(encoded_frame, render_time_ms);
+        }
+        int32_t RegisterDecodeCompleteCallback(DecodedImageCallback* callback) {
+          return decoder_->RegisterDecodeCompleteCallback(callback);
+        }
+        TestVideoDecoder* decoder_;
+      };
+      decoders_.push_back(std::make_unique<NiceMock<TestVideoDecoder>>());
+      return std::make_unique<NiceMock<DecoderWrapper>>(decoders_.back().get());
+    }));
 
     int num_spatial_layers =
         ScalabilityModeToNumSpatialLayers(scalability_mode);
@@ -252,7 +250,7 @@ class VideoCodecTesterTest : public ::testing::Test {
 
     std::unique_ptr<VideoCodecStats> stats =
         VideoCodecTester::RunEncodeDecodeTest(
-            video_source_settings, &encoder_factory, &decoder_factory,
+            env_, video_source_settings, &encoder_factory, &decoder_factory,
             EncoderSettings{}, DecoderSettings{}, encoding_settings);
 
     remove(yuv_path.c_str());
@@ -260,6 +258,7 @@ class VideoCodecTesterTest : public ::testing::Test {
   }
 
  protected:
+  const Environment env_ = CreateEnvironment();
   std::vector<std::unique_ptr<TestVideoDecoder>> decoders_;
 };
 
@@ -605,6 +604,7 @@ class VideoCodecTesterTestPacing
   void TearDown() override { remove(source_yuv_file_path_.c_str()); }
 
  protected:
+  const Environment env_ = CreateEnvironment();
   std::string source_yuv_file_path_;
 };
 
@@ -644,15 +644,14 @@ TEST_P(VideoCodecTesterTestPacing, PaceDecode) {
   MockCodedVideoSource video_source(kNumFrames, kTargetFramerate);
 
   NiceMock<MockVideoDecoderFactory> decoder_factory;
-  ON_CALL(decoder_factory, CreateVideoDecoder(_))
-      .WillByDefault([](const SdpVideoFormat&) {
-        return std::make_unique<NiceMock<MockVideoDecoder>>();
-      });
+  ON_CALL(decoder_factory, Create).WillByDefault(WithoutArgs([] {
+    return std::make_unique<NiceMock<MockVideoDecoder>>();
+  }));
 
   DecoderSettings decoder_settings;
   decoder_settings.pacing_settings = pacing_settings;
   std::vector<Frame> frames =
-      VideoCodecTester::RunDecodeTest(&video_source, &decoder_factory,
+      VideoCodecTester::RunDecodeTest(env_, &video_source, &decoder_factory,
                                       decoder_settings, SdpVideoFormat("VP8"))
           ->Slice(/*filter=*/{}, /*merge=*/false);
   ASSERT_THAT(frames, SizeIs(kNumFrames));
@@ -676,5 +675,137 @@ INSTANTIATE_TEST_SUITE_P(
         std::make_tuple(PacingSettings{.mode = PacingMode::kConstantRate,
                                        .constant_rate = Frequency::Hertz(20)},
                         /*expected_delta_ms=*/50)));
+
+struct EncodingSettingsTestParameters {
+  std::string codec_type;
+  std::string scalability_mode;
+  std::vector<int> bitrate_kbps;
+  std::vector<int> expected_bitrate_kbps;
+};
+
+class VideoCodecTesterTestEncodingSettings
+    : public ::testing::TestWithParam<EncodingSettingsTestParameters> {};
+
+TEST_P(VideoCodecTesterTestEncodingSettings, CreateEncodingSettings) {
+  EncodingSettingsTestParameters test_params = GetParam();
+  std::map<uint32_t, EncodingSettings> encoding_settings =
+      VideoCodecTester::CreateEncodingSettings(
+          test_params.codec_type, test_params.scalability_mode, /*width=*/1280,
+          /*height=*/720, test_params.bitrate_kbps, /*framerate_fps=*/30,
+          /*num_frames=*/1);
+  ASSERT_THAT(encoding_settings, SizeIs(1));
+  const std::map<LayerId, LayerSettings>& layers_settings =
+      encoding_settings.begin()->second.layers_settings;
+  std::vector<int> configured_bitrate_kbps;
+  std::transform(layers_settings.begin(), layers_settings.end(),
+                 std::back_inserter(configured_bitrate_kbps),
+                 [](const auto& layer_settings) {
+                   return layer_settings.second.bitrate.kbps();
+                 });
+  EXPECT_EQ(configured_bitrate_kbps, test_params.expected_bitrate_kbps);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    Vp8,
+    VideoCodecTesterTestEncodingSettings,
+    Values(EncodingSettingsTestParameters{.codec_type = "VP8",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {1},
+                                          .expected_bitrate_kbps = {1}},
+           EncodingSettingsTestParameters{.codec_type = "VP8",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {10000},
+                                          .expected_bitrate_kbps = {10000}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP8",
+               .scalability_mode = "L1T3",
+               .bitrate_kbps = {1000},
+               .expected_bitrate_kbps = {400, 200, 400}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP8",
+               .scalability_mode = "S3T3",
+               .bitrate_kbps = {100},
+               .expected_bitrate_kbps = {40, 20, 40, 0, 0, 0, 0, 0, 0}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP8",
+               .scalability_mode = "S3T3",
+               .bitrate_kbps = {10000},
+               .expected_bitrate_kbps = {60, 30, 60, 200, 100, 200, 1000, 500,
+                                         1000}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP8",
+               .scalability_mode = "S3T3",
+               .bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800, 900},
+               .expected_bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800,
+                                         900}}));
+
+INSTANTIATE_TEST_SUITE_P(
+    Vp9,
+    VideoCodecTesterTestEncodingSettings,
+    Values(EncodingSettingsTestParameters{.codec_type = "VP9",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {1},
+                                          .expected_bitrate_kbps = {1}},
+           EncodingSettingsTestParameters{.codec_type = "VP9",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {10000},
+                                          .expected_bitrate_kbps = {10000}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP9",
+               .scalability_mode = "L1T3",
+               .bitrate_kbps = {1000},
+               .expected_bitrate_kbps = {540, 163, 297}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP9",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {100},
+               .expected_bitrate_kbps = {54, 16, 30, 0, 0, 0, 0, 0, 0}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP9",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {10000},
+               .expected_bitrate_kbps = {77, 23, 42, 226, 68, 124, 823, 249,
+                                         452}},
+           EncodingSettingsTestParameters{
+               .codec_type = "VP9",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800, 900},
+               .expected_bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800,
+                                         900}}));
+
+INSTANTIATE_TEST_SUITE_P(
+    Av1,
+    VideoCodecTesterTestEncodingSettings,
+    Values(EncodingSettingsTestParameters{.codec_type = "AV1",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {1},
+                                          .expected_bitrate_kbps = {1}},
+           EncodingSettingsTestParameters{.codec_type = "AV1",
+                                          .scalability_mode = "L1T1",
+                                          .bitrate_kbps = {10000},
+                                          .expected_bitrate_kbps = {10000}},
+           EncodingSettingsTestParameters{
+               .codec_type = "AV1",
+               .scalability_mode = "L1T3",
+               .bitrate_kbps = {1000},
+               .expected_bitrate_kbps = {540, 163, 297}},
+           EncodingSettingsTestParameters{
+               .codec_type = "AV1",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {100},
+               .expected_bitrate_kbps = {54, 16, 30, 0, 0, 0, 0, 0, 0}},
+           EncodingSettingsTestParameters{
+               .codec_type = "AV1",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {10000},
+               .expected_bitrate_kbps = {77, 23, 42, 226, 68, 124, 823, 249,
+                                         452}},
+           EncodingSettingsTestParameters{
+               .codec_type = "AV1",
+               .scalability_mode = "L3T3",
+               .bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800, 900},
+               .expected_bitrate_kbps = {100, 200, 300, 400, 500, 600, 700, 800,
+                                         900}}));
+
 }  // namespace test
 }  // namespace webrtc