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-rw-r--r--third_party/libwebrtc/test/BUILD.gn55
-rw-r--r--third_party/libwebrtc/test/call_test.cc72
-rw-r--r--third_party/libwebrtc/test/call_test.h17
-rw-r--r--third_party/libwebrtc/test/fuzzers/BUILD.gn1
-rw-r--r--third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc6
-rw-r--r--third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc2
-rw-r--r--third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc3
-rw-r--r--third_party/libwebrtc/test/fuzzers/utils/BUILD.gn4
-rw-r--r--third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc11
-rw-r--r--third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h1
-rw-r--r--third_party/libwebrtc/test/network/BUILD.gn3
-rw-r--r--third_party/libwebrtc/test/network/emulated_turn_server.cc102
-rw-r--r--third_party/libwebrtc/test/network/emulated_turn_server.h3
-rw-r--r--third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc14
-rw-r--r--third_party/libwebrtc/test/pc/e2e/BUILD.gn3
-rw-r--r--third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc50
-rw-r--r--third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc24
-rw-r--r--third_party/libwebrtc/test/peer_scenario/scenario_connection.cc20
-rw-r--r--third_party/libwebrtc/test/rtp_test_utils_gn/moz.build7
-rw-r--r--third_party/libwebrtc/test/scenario/BUILD.gn4
-rw-r--r--third_party/libwebrtc/test/scenario/audio_stream.cc2
-rw-r--r--third_party/libwebrtc/test/scenario/call_client.cc53
-rw-r--r--third_party/libwebrtc/test/scenario/call_client.h6
-rw-r--r--third_party/libwebrtc/test/scenario/scenario_config.h1
-rw-r--r--third_party/libwebrtc/test/scenario/video_stream.cc18
-rw-r--r--third_party/libwebrtc/test/video_codec_tester.cc1324
-rw-r--r--third_party/libwebrtc/test/video_codec_tester.h227
-rw-r--r--third_party/libwebrtc/test/video_codec_tester_unittest.cc513
28 files changed, 2312 insertions, 234 deletions
diff --git a/third_party/libwebrtc/test/BUILD.gn b/third_party/libwebrtc/test/BUILD.gn
index f7980f941d..854530c01e 100644
--- a/third_party/libwebrtc/test/BUILD.gn
+++ b/third_party/libwebrtc/test/BUILD.gn
@@ -724,6 +724,7 @@ if (rtc_include_tests) {
":test_main",
":test_support",
":test_support_test_artifacts",
+ ":video_codec_tester",
":video_test_common",
":video_test_support",
":y4m_frame_generator",
@@ -731,11 +732,15 @@ if (rtc_include_tests) {
"../api:create_frame_generator",
"../api:create_simulcast_test_fixture_api",
"../api:frame_generator_api",
+ "../api:mock_video_codec_factory",
+ "../api:mock_video_decoder",
+ "../api:mock_video_encoder",
"../api:scoped_refptr",
"../api:simulcast_test_fixture_api",
"../api/task_queue:task_queue_test",
"../api/test/video:function_video_factory",
"../api/test/video:video_frame_writer",
+ "../api/units:data_rate",
"../api/units:time_delta",
"../api/video:encoded_image",
"../api/video:video_frame",
@@ -752,6 +757,7 @@ if (rtc_include_tests) {
"../modules/video_coding:webrtc_h264",
"../modules/video_coding:webrtc_vp8",
"../modules/video_coding:webrtc_vp9",
+ "../modules/video_coding/svc:scalability_mode_util",
"../rtc_base:criticalsection",
"../rtc_base:rtc_event",
"../rtc_base:rtc_task_queue",
@@ -765,6 +771,7 @@ if (rtc_include_tests) {
"scenario:scenario_unittests",
"time_controller:time_controller",
"time_controller:time_controller_unittests",
+ "//third_party/libyuv",
]
absl_deps = [
"//third_party/abseil-cpp/absl/flags:flag",
@@ -789,6 +796,7 @@ if (rtc_include_tests) {
"testsupport/y4m_frame_writer_unittest.cc",
"testsupport/yuv_frame_reader_unittest.cc",
"testsupport/yuv_frame_writer_unittest.cc",
+ "video_codec_tester_unittest.cc",
]
if (rtc_enable_protobuf) {
@@ -1287,9 +1295,10 @@ if (!build_with_chromium) {
"../api:transport_api",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/environment",
+ "../api/environment:environment_factory",
"../api/rtc_event_log",
"../api/task_queue",
- "../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/transport:field_trial_based_config",
"../api/units:time_delta",
@@ -1378,3 +1387,47 @@ rtc_library("fake_encoded_frame") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+
+rtc_library("video_codec_tester") {
+ testonly = true
+ sources = [
+ "video_codec_tester.cc",
+ "video_codec_tester.h",
+ ]
+ deps = [
+ "../api:array_view",
+ "../api/numerics:numerics",
+ "../api/test/metrics:metric",
+ "../api/test/metrics:metrics_logger",
+ "../api/units:data_rate",
+ "../api/units:data_size",
+ "../api/units:frequency",
+ "../api/units:time_delta",
+ "../api/units:timestamp",
+ "../api/video:builtin_video_bitrate_allocator_factory",
+ "../api/video:encoded_image",
+ "../api/video:resolution",
+ "../api/video:video_bitrate_allocator",
+ "../api/video:video_frame",
+ "../api/video_codecs:video_codecs_api",
+ "../media:media_constants",
+ "../modules/video_coding:video_codec_interface",
+ "../modules/video_coding:video_coding_utility",
+ "../modules/video_coding:webrtc_vp9_helpers",
+ "../modules/video_coding/codecs/av1:av1_svc_config",
+ "../modules/video_coding/svc:scalability_mode_util",
+ "../rtc_base:checks",
+ "../rtc_base:logging",
+ "../rtc_base:rtc_event",
+ "../rtc_base:stringutils",
+ "../rtc_base:task_queue_for_test",
+ "../rtc_base:timeutils",
+ "../rtc_base/synchronization:mutex",
+ "../system_wrappers",
+ "../test:fileutils",
+ "../test:video_test_support",
+ "//third_party/libyuv",
+ ]
+
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
diff --git a/third_party/libwebrtc/test/call_test.cc b/third_party/libwebrtc/test/call_test.cc
index 2b71608ac6..09099cccd6 100644
--- a/third_party/libwebrtc/test/call_test.cc
+++ b/third_party/libwebrtc/test/call_test.cc
@@ -15,7 +15,8 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/task_queue/default_task_queue_factory.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/create_frame_generator.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
@@ -38,19 +39,18 @@ namespace webrtc {
namespace test {
CallTest::CallTest()
- : clock_(Clock::GetRealTimeClock()),
- task_queue_factory_(CreateDefaultTaskQueueFactory()),
- send_event_log_(std::make_unique<RtcEventLogNull>()),
- recv_event_log_(std::make_unique<RtcEventLogNull>()),
+ : env_(CreateEnvironment(&field_trials_)),
+ send_env_(env_),
+ recv_env_(env_),
audio_send_config_(/*send_transport=*/nullptr),
audio_send_stream_(nullptr),
frame_generator_capturer_(nullptr),
fake_encoder_factory_([this]() {
std::unique_ptr<FakeEncoder> fake_encoder;
if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
- fake_encoder = std::make_unique<FakeVp8Encoder>(clock_);
+ fake_encoder = std::make_unique<FakeVp8Encoder>(&env_.clock());
} else {
- fake_encoder = std::make_unique<FakeEncoder>(clock_);
+ fake_encoder = std::make_unique<FakeEncoder>(&env_.clock());
}
fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
return fake_encoder;
@@ -62,12 +62,24 @@ CallTest::CallTest()
num_flexfec_streams_(0),
audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
- task_queue_(task_queue_factory_->CreateTaskQueue(
+ task_queue_(env_.task_queue_factory().CreateTaskQueue(
"CallTestTaskQueue",
TaskQueueFactory::Priority::NORMAL)) {}
CallTest::~CallTest() = default;
+void CallTest::SetSendEventLog(std::unique_ptr<RtcEventLog> event_log) {
+ EnvironmentFactory f(env_);
+ f.Set(std::move(event_log));
+ send_env_ = f.Create();
+}
+
+void CallTest::SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log) {
+ EnvironmentFactory f(env_);
+ f.Set(std::move(event_log));
+ recv_env_ = f.Create();
+}
+
void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
for (const RtpExtension& registered_extension : rtp_extensions_) {
if (registered_extension.id == extension.id) {
@@ -97,7 +109,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
num_audio_streams_ = test->GetNumAudioStreams();
num_flexfec_streams_ = test->GetNumFlexfecStreams();
RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
- CallConfig send_config(send_event_log_.get());
+ CallConfig send_config = SendCallConfig();
test->ModifySenderBitrateConfig(&send_config.bitrate_config);
if (num_audio_streams_ > 0) {
CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
@@ -117,7 +129,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
}
CreateSenderCall(send_config);
if (test->ShouldCreateReceivers()) {
- CallConfig recv_config(recv_event_log_.get());
+ CallConfig recv_config = RecvCallConfig();
test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
if (num_audio_streams_ > 0) {
AudioState::Config audio_state_config;
@@ -206,9 +218,20 @@ void CallTest::RunBaseTest(BaseTest* test) {
});
}
+CallConfig CallTest::SendCallConfig() const {
+ CallConfig sender_config(send_env_);
+ sender_config.network_state_predictor_factory =
+ network_state_predictor_factory_.get();
+ sender_config.network_controller_factory = network_controller_factory_.get();
+ return sender_config;
+}
+
+CallConfig CallTest::RecvCallConfig() const {
+ return CallConfig(recv_env_);
+}
+
void CallTest::CreateCalls() {
- CreateCalls(CallConfig(send_event_log_.get()),
- CallConfig(recv_event_log_.get()));
+ CreateCalls(SendCallConfig(), RecvCallConfig());
}
void CallTest::CreateCalls(const CallConfig& sender_config,
@@ -218,24 +241,15 @@ void CallTest::CreateCalls(const CallConfig& sender_config,
}
void CallTest::CreateSenderCall() {
- CreateSenderCall(CallConfig(send_event_log_.get()));
+ CreateSenderCall(SendCallConfig());
}
void CallTest::CreateSenderCall(const CallConfig& config) {
- auto sender_config = config;
- sender_config.task_queue_factory = task_queue_factory_.get();
- sender_config.network_state_predictor_factory =
- network_state_predictor_factory_.get();
- sender_config.network_controller_factory = network_controller_factory_.get();
- sender_config.trials = &field_trials_;
- sender_call_ = Call::Create(sender_config);
+ sender_call_ = Call::Create(config);
}
void CallTest::CreateReceiverCall(const CallConfig& config) {
- auto receiver_config = config;
- receiver_config.task_queue_factory = task_queue_factory_.get();
- receiver_config.trials = &field_trials_;
- receiver_call_ = Call::Create(receiver_config);
+ receiver_call_ = Call::Create(config);
}
void CallTest::DestroyCalls() {
@@ -489,7 +503,7 @@ void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
clock,
test::CreateSquareFrameGenerator(width, height, absl::nullopt,
absl::nullopt),
- framerate * speed, *task_queue_factory_);
+ framerate * speed, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
@@ -502,10 +516,10 @@ void CallTest::CreateFrameGeneratorCapturer(int framerate,
video_sources_.clear();
auto frame_generator_capturer =
std::make_unique<test::FrameGeneratorCapturer>(
- clock_,
+ &env_.clock(),
test::CreateSquareFrameGenerator(width, height, absl::nullopt,
absl::nullopt),
- framerate, *task_queue_factory_);
+ framerate, env_.task_queue_factory());
frame_generator_capturer_ = frame_generator_capturer.get();
frame_generator_capturer->Init();
video_sources_.push_back(std::move(frame_generator_capturer));
@@ -516,9 +530,9 @@ void CallTest::CreateFakeAudioDevices(
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
fake_send_audio_device_ = TestAudioDeviceModule::Create(
- task_queue_factory_.get(), std::move(capturer), nullptr, 1.f);
+ &env_.task_queue_factory(), std::move(capturer), nullptr, 1.f);
fake_recv_audio_device_ = TestAudioDeviceModule::Create(
- task_queue_factory_.get(), nullptr, std::move(renderer), 1.f);
+ &env_.task_queue_factory(), nullptr, std::move(renderer), 1.f);
}
void CallTest::CreateVideoStreams() {
diff --git a/third_party/libwebrtc/test/call_test.h b/third_party/libwebrtc/test/call_test.h
index 8d2b001f72..decf02f20b 100644
--- a/third_party/libwebrtc/test/call_test.h
+++ b/third_party/libwebrtc/test/call_test.h
@@ -17,6 +17,7 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
+#include "api/environment/environment.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
@@ -52,6 +53,11 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
static const std::map<uint8_t, MediaType> payload_type_map_;
protected:
+ const Environment& env() const { return env_; }
+
+ void SetSendEventLog(std::unique_ptr<RtcEventLog> event_log);
+ void SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log);
+
void RegisterRtpExtension(const RtpExtension& extension);
// Returns header extensions that can be parsed by the transport.
rtc::ArrayView<const RtpExtension> GetRegisteredExtensions() {
@@ -62,6 +68,9 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
// to simplify test code.
void RunBaseTest(BaseTest* test);
+ CallConfig SendCallConfig() const;
+ CallConfig RecvCallConfig() const;
+
void CreateCalls();
void CreateCalls(const CallConfig& sender_config,
const CallConfig& receiver_config);
@@ -185,13 +194,11 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
void OnRtpPacket(const RtpPacketReceived& packet) override;
test::RunLoop loop_;
-
- Clock* const clock_;
test::ScopedKeyValueConfig field_trials_;
+ Environment env_;
+ Environment send_env_;
+ Environment recv_env_;
- std::unique_ptr<TaskQueueFactory> task_queue_factory_;
- std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
- std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
std::unique_ptr<Call> sender_call_;
std::unique_ptr<PacketTransport> send_transport_;
SimulatedNetworkInterface* send_simulated_network_ = nullptr;
diff --git a/third_party/libwebrtc/test/fuzzers/BUILD.gn b/third_party/libwebrtc/test/fuzzers/BUILD.gn
index 3d2ac7a7e7..43e9a5e922 100644
--- a/third_party/libwebrtc/test/fuzzers/BUILD.gn
+++ b/third_party/libwebrtc/test/fuzzers/BUILD.gn
@@ -12,6 +12,7 @@ import("../../webrtc.gni")
rtc_library("webrtc_fuzzer_main") {
sources = [ "webrtc_fuzzer_main.cc" ]
+ testonly = true
deps = [
"../../rtc_base:logging",
"//testing/libfuzzer:fuzzing_engine_main",
diff --git a/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc
index 04a459bc71..87b76858bb 100644
--- a/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc
@@ -34,7 +34,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
ForwardErrorCorrection::CreateFlexfec(kFecSsrc, kMediaSsrc);
// Entropy from fuzzer.
- rtc::ByteBufferReader fuzz_buffer(reinterpret_cast<const char*>(data), size);
+ rtc::ByteBufferReader fuzz_buffer(rtc::MakeArrayView(data, size));
// Initial stream state.
uint16_t media_seqnum;
@@ -75,8 +75,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
uint8_t packet_type;
uint8_t packet_loss;
while (true) {
- if (!fuzz_buffer.ReadBytes(reinterpret_cast<char*>(packet_buffer),
- kPacketSize)) {
+ if (!fuzz_buffer.ReadBytes(
+ rtc::ArrayView<uint8_t>(packet_buffer, kPacketSize))) {
return;
}
if (!fuzz_buffer.ReadUInt8(&reordering))
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
index ea66c4a1c8..93673f0b8e 100644
--- a/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
@@ -23,7 +23,7 @@ class DataReader {
template <typename T>
void CopyTo(T* object) {
- static_assert(std::is_pod<T>(), "");
+ static_assert(std::is_trivial_v<T> && std::is_standard_layout_v<T>, "");
uint8_t* destination = reinterpret_cast<uint8_t*>(object);
size_t object_size = sizeof(T);
size_t num_bytes = std::min(size_ - offset_, object_size);
diff --git a/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc
index 6ca9eac8b2..c8aad8c232 100644
--- a/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc
+++ b/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc
@@ -15,14 +15,13 @@
namespace webrtc {
void FuzzOneInput(const uint8_t* data, size_t size) {
- const char* message = reinterpret_cast<const char*>(data);
// Normally we'd check the integrity first, but those checks are
// fuzzed separately in stun_validator_fuzzer.cc. We still want to
// fuzz this target since the integrity checks could be forged by a
// malicious adversary who receives a call.
std::unique_ptr<cricket::IceMessage> stun_msg(new cricket::IceMessage());
- rtc::ByteBufferReader buf(message, size);
+ rtc::ByteBufferReader buf(rtc::MakeArrayView(data, size));
stun_msg->Read(&buf);
stun_msg->ValidateMessageIntegrity("");
}
diff --git a/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn b/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn
index dfb617857c..895e76e939 100644
--- a/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn
+++ b/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn
@@ -15,10 +15,8 @@ rtc_library("rtp_replayer") {
"rtp_replayer.h",
]
deps = [
- "../../../api/rtc_event_log",
- "../../../api/task_queue:default_task_queue_factory",
+ "../../../api/environment:environment_factory",
"../../../api/test/video:function_video_factory",
- "../../../api/transport:field_trial_based_config",
"../../../api/units:timestamp",
"../../../api/video_codecs:video_codecs_api",
"../../../call",
diff --git a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc
index f6d7119338..1b1b96803f 100644
--- a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc
+++ b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc
@@ -16,8 +16,7 @@
#include <utility>
#include "absl/memory/memory.h"
-#include "api/task_queue/default_task_queue_factory.h"
-#include "api/transport/field_trial_based_config.h"
+#include "api/environment/environment_factory.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -73,13 +72,7 @@ void RtpReplayer::Replay(
}
// Setup the video streams based on the configuration.
- webrtc::RtcEventLogNull event_log;
- std::unique_ptr<TaskQueueFactory> task_queue_factory =
- CreateDefaultTaskQueueFactory();
- CallConfig call_config(&event_log);
- call_config.task_queue_factory = task_queue_factory.get();
- FieldTrialBasedConfig field_trials;
- call_config.trials = &field_trials;
+ CallConfig call_config(CreateEnvironment());
std::unique_ptr<Call> call(Call::Create(call_config));
SetupVideoStreams(&receive_stream_configs, stream_state.get(), call.get());
diff --git a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
index ae94a640a5..73220eecfa 100644
--- a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
+++ b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
@@ -18,7 +18,6 @@
#include <string>
#include <vector>
-#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/video_codecs/video_decoder.h"
#include "call/call.h"
diff --git a/third_party/libwebrtc/test/network/BUILD.gn b/third_party/libwebrtc/test/network/BUILD.gn
index 5a6cb31f4b..b8255d35fd 100644
--- a/third_party/libwebrtc/test/network/BUILD.gn
+++ b/third_party/libwebrtc/test/network/BUILD.gn
@@ -76,6 +76,7 @@ rtc_library("emulated_network") {
"../../rtc_base:task_queue_for_test",
"../../rtc_base:threading",
"../../rtc_base/memory:always_valid_pointer",
+ "../../rtc_base/network:received_packet",
"../../rtc_base/synchronization:mutex",
"../../rtc_base/system:no_unique_address",
"../../rtc_base/task_utils:repeating_task",
@@ -117,6 +118,7 @@ if (rtc_include_tests && !build_with_chromium) {
":emulated_network",
"../:test_support",
"../../api:callfactory_api",
+ "../../api:enable_media_with_defaults",
"../../api:libjingle_peerconnection_api",
"../../api:scoped_refptr",
"../../api:simulated_network_api",
@@ -125,7 +127,6 @@ if (rtc_include_tests && !build_with_chromium) {
"../../api/transport:field_trial_based_config",
"../../call:simulated_network",
"../../media:rtc_audio_video",
- "../../media:rtc_media_engine_defaults",
"../../modules/audio_device:test_audio_device_module",
"../../p2p:rtc_p2p",
"../../pc:pc_test_utils",
diff --git a/third_party/libwebrtc/test/network/emulated_turn_server.cc b/third_party/libwebrtc/test/network/emulated_turn_server.cc
index 0bc7ec6e2a..93724ca8a3 100644
--- a/third_party/libwebrtc/test/network/emulated_turn_server.cc
+++ b/third_party/libwebrtc/test/network/emulated_turn_server.cc
@@ -14,6 +14,7 @@
#include <utility>
#include "api/packet_socket_factory.h"
+#include "rtc_base/network/received_packet.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/task_queue_for_test.h"
@@ -22,10 +23,56 @@ namespace {
static const char kTestRealm[] = "example.org";
static const char kTestSoftware[] = "TestTurnServer";
+// A wrapper class for cricket::TurnServer to allocate sockets.
+class PacketSocketFactoryWrapper : public rtc::PacketSocketFactory {
+ public:
+ explicit PacketSocketFactoryWrapper(
+ webrtc::test::EmulatedTURNServer* turn_server)
+ : turn_server_(turn_server) {}
+ ~PacketSocketFactoryWrapper() override {}
+
+ // This method is called from TurnServer when making a TURN ALLOCATION.
+ // It will create a socket on the `peer_` endpoint.
+ rtc::AsyncPacketSocket* CreateUdpSocket(const rtc::SocketAddress& address,
+ uint16_t min_port,
+ uint16_t max_port) override {
+ return turn_server_->CreatePeerSocket();
+ }
+
+ rtc::AsyncListenSocket* CreateServerTcpSocket(
+ const rtc::SocketAddress& local_address,
+ uint16_t min_port,
+ uint16_t max_port,
+ int opts) override {
+ return nullptr;
+ }
+ rtc::AsyncPacketSocket* CreateClientTcpSocket(
+ const rtc::SocketAddress& local_address,
+ const rtc::SocketAddress& remote_address,
+ const rtc::ProxyInfo& proxy_info,
+ const std::string& user_agent,
+ const rtc::PacketSocketTcpOptions& tcp_options) override {
+ return nullptr;
+ }
+ std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAsyncDnsResolver()
+ override {
+ return nullptr;
+ }
+
+ private:
+ webrtc::test::EmulatedTURNServer* turn_server_;
+};
+
+} // namespace
+
+namespace webrtc {
+namespace test {
+
// A wrapper class for copying data between an AsyncPacketSocket and a
// EmulatedEndpoint. This is used by the cricket::TurnServer when
// sending data back into the emulated network.
-class AsyncPacketSocketWrapper : public rtc::AsyncPacketSocket {
+class EmulatedTURNServer::AsyncPacketSocketWrapper
+ : public rtc::AsyncPacketSocket {
public:
AsyncPacketSocketWrapper(webrtc::test::EmulatedTURNServer* turn_server,
webrtc::EmulatedEndpoint* endpoint,
@@ -56,6 +103,9 @@ class AsyncPacketSocketWrapper : public rtc::AsyncPacketSocket {
return cb;
}
int Close() override { return 0; }
+ void NotifyPacketReceived(const rtc::ReceivedPacket& packet) {
+ rtc::AsyncPacketSocket::NotifyPacketReceived(packet);
+ }
rtc::AsyncPacketSocket::State GetState() const override {
return rtc::AsyncPacketSocket::STATE_BOUND;
@@ -71,51 +121,6 @@ class AsyncPacketSocketWrapper : public rtc::AsyncPacketSocket {
const rtc::SocketAddress local_address_;
};
-// A wrapper class for cricket::TurnServer to allocate sockets.
-class PacketSocketFactoryWrapper : public rtc::PacketSocketFactory {
- public:
- explicit PacketSocketFactoryWrapper(
- webrtc::test::EmulatedTURNServer* turn_server)
- : turn_server_(turn_server) {}
- ~PacketSocketFactoryWrapper() override {}
-
- // This method is called from TurnServer when making a TURN ALLOCATION.
- // It will create a socket on the `peer_` endpoint.
- rtc::AsyncPacketSocket* CreateUdpSocket(const rtc::SocketAddress& address,
- uint16_t min_port,
- uint16_t max_port) override {
- return turn_server_->CreatePeerSocket();
- }
-
- rtc::AsyncListenSocket* CreateServerTcpSocket(
- const rtc::SocketAddress& local_address,
- uint16_t min_port,
- uint16_t max_port,
- int opts) override {
- return nullptr;
- }
- rtc::AsyncPacketSocket* CreateClientTcpSocket(
- const rtc::SocketAddress& local_address,
- const rtc::SocketAddress& remote_address,
- const rtc::ProxyInfo& proxy_info,
- const std::string& user_agent,
- const rtc::PacketSocketTcpOptions& tcp_options) override {
- return nullptr;
- }
- std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAsyncDnsResolver()
- override {
- return nullptr;
- }
-
- private:
- webrtc::test::EmulatedTURNServer* turn_server_;
-};
-
-} // namespace
-
-namespace webrtc {
-namespace test {
-
EmulatedTURNServer::EmulatedTURNServer(std::unique_ptr<rtc::Thread> thread,
EmulatedEndpoint* client,
EmulatedEndpoint* peer)
@@ -170,9 +175,8 @@ void EmulatedTURNServer::OnPacketReceived(webrtc::EmulatedIpPacket packet) {
RTC_DCHECK_RUN_ON(thread_.get());
auto it = sockets_.find(packet.to);
if (it != sockets_.end()) {
- it->second->SignalReadPacket(
- it->second, reinterpret_cast<const char*>(packet.cdata()),
- packet.size(), packet.from, packet.arrival_time.ms());
+ it->second->NotifyPacketReceived(
+ rtc::ReceivedPacket(packet.data, packet.from, packet.arrival_time));
}
});
}
diff --git a/third_party/libwebrtc/test/network/emulated_turn_server.h b/third_party/libwebrtc/test/network/emulated_turn_server.h
index 9cb0ceabf6..de5d266897 100644
--- a/third_party/libwebrtc/test/network/emulated_turn_server.h
+++ b/third_party/libwebrtc/test/network/emulated_turn_server.h
@@ -84,7 +84,8 @@ class EmulatedTURNServer : public EmulatedTURNServerInterface,
EmulatedEndpoint* const client_;
EmulatedEndpoint* const peer_;
std::unique_ptr<cricket::TurnServer> turn_server_ RTC_GUARDED_BY(&thread_);
- std::map<rtc::SocketAddress, rtc::AsyncPacketSocket*> sockets_
+ class AsyncPacketSocketWrapper;
+ std::map<rtc::SocketAddress, AsyncPacketSocketWrapper*> sockets_
RTC_GUARDED_BY(&thread_);
// Wraps a EmulatedEndpoint in a AsyncPacketSocket to bridge interaction
diff --git a/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc b/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc
index 51a45a8234..09d3946747 100644
--- a/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc
+++ b/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc
@@ -11,7 +11,7 @@
#include <cstdint>
#include <memory>
-#include "api/call/call_factory_interface.h"
+#include "api/enable_media_with_defaults.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/scoped_refptr.h"
@@ -19,7 +19,6 @@
#include "api/transport/field_trial_based_config.h"
#include "call/simulated_network.h"
#include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "p2p/client/basic_port_allocator.h"
@@ -57,23 +56,18 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
rtc::Thread* network_thread) {
PeerConnectionFactoryDependencies pcf_deps;
pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
- pcf_deps.call_factory = CreateCallFactory();
pcf_deps.event_log_factory =
std::make_unique<RtcEventLogFactory>(pcf_deps.task_queue_factory.get());
pcf_deps.network_thread = network_thread;
pcf_deps.signaling_thread = signaling_thread;
pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
- cricket::MediaEngineDependencies media_deps;
- media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
- media_deps.adm = TestAudioDeviceModule::Create(
- media_deps.task_queue_factory,
+ pcf_deps.adm = TestAudioDeviceModule::Create(
+ pcf_deps.task_queue_factory.get(),
TestAudioDeviceModule::CreatePulsedNoiseCapturer(kMaxAptitude,
kSamplingFrequency),
TestAudioDeviceModule::CreateDiscardRenderer(kSamplingFrequency),
/*speed=*/1.f);
- media_deps.trials = pcf_deps.trials.get();
- SetMediaEngineDefaults(&media_deps);
- pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+ EnableMediaWithDefaults(pcf_deps);
return CreateModularPeerConnectionFactory(std::move(pcf_deps));
}
diff --git a/third_party/libwebrtc/test/pc/e2e/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
index 86a9110f58..9d1c1d437f 100644
--- a/third_party/libwebrtc/test/pc/e2e/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@@ -96,6 +96,7 @@ if (!build_with_chromium) {
":test_peer",
"../..:copy_to_file_audio_capturer",
"../../../api:create_time_controller",
+ "../../../api:enable_media_with_defaults",
"../../../api:time_controller",
"../../../api/rtc_event_log:rtc_event_log_factory",
"../../../api/task_queue:default_task_queue_factory",
@@ -105,8 +106,6 @@ if (!build_with_chromium) {
"../../../api/transport:field_trial_based_config",
"../../../api/video_codecs:builtin_video_decoder_factory",
"../../../api/video_codecs:builtin_video_encoder_factory",
- "../../../media:rtc_audio_video",
- "../../../media:rtc_media_engine_defaults",
"../../../modules/audio_device:test_audio_device_module",
"../../../modules/audio_processing/aec_dump",
"../../../p2p:rtc_p2p",
diff --git a/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
index 41f7533c3a..dd900027ee 100644
--- a/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
@@ -13,6 +13,7 @@
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
+#include "api/enable_media_with_defaults.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/create_time_controller.h"
#include "api/test/pclf/media_configuration.h"
@@ -21,8 +22,6 @@
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
-#include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
#include "p2p/client/basic_port_allocator.h"
#include "rtc_base/thread.h"
@@ -146,28 +145,6 @@ rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
std::move(renderer), /*speed=*/1.f);
}
-std::unique_ptr<cricket::MediaEngineInterface> CreateMediaEngine(
- TaskQueueFactory* task_queue_factory,
- PeerConnectionFactoryComponents* pcf_dependencies,
- rtc::scoped_refptr<AudioDeviceModule> audio_device_module) {
- cricket::MediaEngineDependencies media_deps;
- media_deps.task_queue_factory = task_queue_factory;
- media_deps.adm = audio_device_module;
- media_deps.audio_processing = pcf_dependencies->audio_processing;
- media_deps.audio_mixer = pcf_dependencies->audio_mixer;
- media_deps.video_encoder_factory =
- std::move(pcf_dependencies->video_encoder_factory);
- media_deps.video_decoder_factory =
- std::move(pcf_dependencies->video_decoder_factory);
- media_deps.audio_encoder_factory = pcf_dependencies->audio_encoder_factory;
- media_deps.audio_decoder_factory = pcf_dependencies->audio_decoder_factory;
- webrtc::SetMediaEngineDefaults(&media_deps);
- RTC_DCHECK(pcf_dependencies->trials);
- media_deps.trials = pcf_dependencies->trials.get();
-
- return cricket::CreateMediaEngine(std::move(media_deps));
-}
-
void WrapVideoEncoderFactory(
absl::string_view peer_name,
double bitrate_multiplier,
@@ -206,7 +183,7 @@ void WrapVideoDecoderFactory(
PeerConnectionFactoryDependencies CreatePCFDependencies(
std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies,
TimeController& time_controller,
- std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+ rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
rtc::Thread* network_thread) {
@@ -214,10 +191,7 @@ PeerConnectionFactoryDependencies CreatePCFDependencies(
pcf_deps.signaling_thread = signaling_thread;
pcf_deps.worker_thread = worker_thread;
pcf_deps.network_thread = network_thread;
- pcf_deps.media_engine = std::move(media_engine);
- pcf_deps.call_factory =
- CreateTimeControllerBasedCallFactory(&time_controller);
pcf_deps.event_log_factory = std::move(pcf_dependencies->event_log_factory);
pcf_deps.task_queue_factory = time_controller.CreateTaskQueueFactory();
@@ -236,6 +210,18 @@ PeerConnectionFactoryDependencies CreatePCFDependencies(
pcf_deps.trials = std::move(pcf_dependencies->trials);
}
+ // Media dependencies
+ pcf_deps.adm = std::move(audio_device_module);
+ pcf_deps.audio_processing = pcf_dependencies->audio_processing;
+ pcf_deps.audio_mixer = pcf_dependencies->audio_mixer;
+ pcf_deps.video_encoder_factory =
+ std::move(pcf_dependencies->video_encoder_factory);
+ pcf_deps.video_decoder_factory =
+ std::move(pcf_dependencies->video_decoder_factory);
+ pcf_deps.audio_encoder_factory = pcf_dependencies->audio_encoder_factory;
+ pcf_deps.audio_decoder_factory = pcf_dependencies->audio_decoder_factory;
+ EnableMediaWithDefaultsAndTimeController(time_controller, pcf_deps);
+
return pcf_deps;
}
@@ -324,10 +310,6 @@ std::unique_ptr<TestPeer> TestPeerFactory::CreateTestPeer(
WrapVideoDecoderFactory(params->name.value(),
components->pcf_dependencies.get(),
video_analyzer_helper_);
- std::unique_ptr<cricket::MediaEngineInterface> media_engine =
- CreateMediaEngine(time_controller_.GetTaskQueueFactory(),
- components->pcf_dependencies.get(),
- audio_device_module);
std::unique_ptr<rtc::Thread> owned_worker_thread =
components->worker_thread != nullptr
@@ -343,8 +325,8 @@ std::unique_ptr<TestPeer> TestPeerFactory::CreateTestPeer(
components->pcf_dependencies->audio_processing;
PeerConnectionFactoryDependencies pcf_deps = CreatePCFDependencies(
std::move(components->pcf_dependencies), time_controller_,
- std::move(media_engine), signaling_thread_, components->worker_thread,
- components->network_thread);
+ std::move(audio_device_module), signaling_thread_,
+ components->worker_thread, components->network_thread);
rtc::scoped_refptr<PeerConnectionFactoryInterface> peer_connection_factory =
CreateModularPeerConnectionFactory(std::move(pcf_deps));
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 697bf055a7..60f2ea7f2e 100644
--- a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
@@ -246,17 +246,13 @@ PeerScenarioClient::PeerScenarioClient(
pcf_deps.network_thread = manager->network_thread();
pcf_deps.signaling_thread = signaling_thread_;
pcf_deps.worker_thread = worker_thread_.get();
- pcf_deps.call_factory =
- CreateTimeControllerBasedCallFactory(net->time_controller());
pcf_deps.task_queue_factory =
net->time_controller()->CreateTaskQueueFactory();
pcf_deps.event_log_factory =
std::make_unique<RtcEventLogFactory>(task_queue_factory_);
pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
- cricket::MediaEngineDependencies media_deps;
- media_deps.task_queue_factory = task_queue_factory_;
- media_deps.adm = TestAudioDeviceModule::Create(
+ pcf_deps.adm = TestAudioDeviceModule::Create(
task_queue_factory_,
TestAudioDeviceModule::CreatePulsedNoiseCapturer(
config.audio.pulsed_noise->amplitude *
@@ -264,28 +260,24 @@ PeerScenarioClient::PeerScenarioClient(
config.audio.sample_rate, config.audio.channels),
TestAudioDeviceModule::CreateDiscardRenderer(config.audio.sample_rate));
- media_deps.audio_processing = AudioProcessingBuilder().Create();
if (config.video.use_fake_codecs) {
- media_deps.video_encoder_factory =
- std::make_unique<FakeVideoEncoderFactory>(
- net->time_controller()->GetClock());
- media_deps.video_decoder_factory =
+ pcf_deps.video_encoder_factory = std::make_unique<FakeVideoEncoderFactory>(
+ net->time_controller()->GetClock());
+ pcf_deps.video_decoder_factory =
std::make_unique<FakeVideoDecoderFactory>();
} else {
- media_deps.video_encoder_factory =
+ pcf_deps.video_encoder_factory =
std::make_unique<VideoEncoderFactoryTemplate<
LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>();
- media_deps.video_decoder_factory =
+ pcf_deps.video_decoder_factory =
std::make_unique<VideoDecoderFactoryTemplate<
LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>();
}
- media_deps.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
- media_deps.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
- media_deps.trials = pcf_deps.trials.get();
- pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+ EnableMediaWithDefaultsAndTimeController(*net->time_controller(), pcf_deps);
+
pcf_deps.fec_controller_factory = nullptr;
pcf_deps.network_controller_factory = nullptr;
pcf_deps.network_state_predictor_factory = nullptr;
diff --git a/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc b/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc
index 66eca275d1..8b2081a4c3 100644
--- a/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc
+++ b/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc
@@ -173,21 +173,14 @@ void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type,
});
auto res = jsep_controller_->SetRemoteDescription(
- remote_description_->GetType(), remote_description_->description());
+ remote_description_->GetType(),
+ local_description_ ? local_description_->description() : nullptr,
+ remote_description_->description());
RTC_CHECK(res.ok()) << res.message();
RtpDemuxerCriteria criteria;
for (const auto& content : remote_description_->description()->contents()) {
- if (content.media_description()->as_audio()) {
- for (const auto& codec :
- content.media_description()->as_audio()->codecs()) {
- criteria.payload_types().insert(codec.id);
- }
- }
- if (content.media_description()->as_video()) {
- for (const auto& codec :
- content.media_description()->as_video()->codecs()) {
- criteria.payload_types().insert(codec.id);
- }
+ for (const auto& codec : content.media_description()->codecs()) {
+ criteria.payload_types().insert(codec.id);
}
}
@@ -203,7 +196,8 @@ void ScenarioIceConnectionImpl::SetLocalSdp(SdpType type,
RTC_DCHECK_RUN_ON(signaling_thread_);
local_description_ = webrtc::CreateSessionDescription(type, local_sdp);
auto res = jsep_controller_->SetLocalDescription(
- local_description_->GetType(), local_description_->description());
+ local_description_->GetType(), local_description_->description(),
+ remote_description_ ? remote_description_->description() : nullptr);
RTC_CHECK(res.ok()) << res.message();
jsep_controller_->MaybeStartGathering();
}
diff --git a/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build b/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build
index f3ffb448cd..5f400c69f9 100644
--- a/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build
+++ b/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build
@@ -192,16 +192,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
OS_LIBS += [
- "android_support",
"unwind"
]
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
- OS_LIBS += [
- "android_support"
- ]
-
if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/test/scenario/BUILD.gn b/third_party/libwebrtc/test/scenario/BUILD.gn
index c3b8847fd1..7ac06c6822 100644
--- a/third_party/libwebrtc/test/scenario/BUILD.gn
+++ b/third_party/libwebrtc/test/scenario/BUILD.gn
@@ -86,10 +86,11 @@ if (rtc_include_tests && !build_with_chromium) {
"../../api:rtp_parameters",
"../../api:sequence_checker",
"../../api:time_controller",
- "../../api:time_controller",
"../../api:transport_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
+ "../../api/environment",
+ "../../api/environment:environment_factory",
"../../api/rtc_event_log",
"../../api/rtc_event_log:rtc_event_log_factory",
"../../api/task_queue",
@@ -146,7 +147,6 @@ if (rtc_include_tests && !build_with_chromium) {
"../../rtc_base/synchronization:mutex",
"../../rtc_base/task_utils:repeating_task",
"../../system_wrappers",
- "../../system_wrappers:field_trial",
"../../video/config:streams_config",
"../logging:log_writer",
"../network:emulated_network",
diff --git a/third_party/libwebrtc/test/scenario/audio_stream.cc b/third_party/libwebrtc/test/scenario/audio_stream.cc
index 7715555e23..5f7db7acdf 100644
--- a/third_party/libwebrtc/test/scenario/audio_stream.cc
+++ b/third_party/libwebrtc/test/scenario/audio_stream.cc
@@ -14,13 +14,11 @@
#include "test/video_test_constants.h"
#if WEBRTC_ENABLE_PROTOBUF
-RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
#else
#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
#endif
-RTC_POP_IGNORING_WUNDEF()
#endif
namespace webrtc {
diff --git a/third_party/libwebrtc/test/scenario/call_client.cc b/third_party/libwebrtc/test/scenario/call_client.cc
index fdf36dee08..8845ad6b0f 100644
--- a/third_party/libwebrtc/test/scenario/call_client.cc
+++ b/third_party/libwebrtc/test/scenario/call_client.cc
@@ -13,6 +13,8 @@
#include <memory>
#include <utility>
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
#include "api/media_types.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
@@ -60,38 +62,27 @@ CallClientFakeAudio InitAudio(TimeController* time_controller) {
}
std::unique_ptr<Call> CreateCall(
- TimeController* time_controller,
- RtcEventLog* event_log,
+ const Environment& env,
CallClientConfig config,
LoggingNetworkControllerFactory* network_controller_factory,
rtc::scoped_refptr<AudioState> audio_state) {
- CallConfig call_config(event_log);
+ CallConfig call_config(env);
call_config.bitrate_config.max_bitrate_bps =
config.transport.rates.max_rate.bps_or(-1);
call_config.bitrate_config.min_bitrate_bps =
config.transport.rates.min_rate.bps();
call_config.bitrate_config.start_bitrate_bps =
config.transport.rates.start_rate.bps();
- call_config.task_queue_factory = time_controller->GetTaskQueueFactory();
call_config.network_controller_factory = network_controller_factory;
call_config.audio_state = audio_state;
- call_config.pacer_burst_interval = config.pacer_burst_interval;
- call_config.trials = config.field_trials;
- Clock* clock = time_controller->GetClock();
- return Call::Create(call_config, clock,
- RtpTransportControllerSendFactory().Create(
- call_config.ExtractTransportConfig(), clock));
+ return Call::Create(call_config);
}
std::unique_ptr<RtcEventLog> CreateEventLog(
- TaskQueueFactory* task_queue_factory,
- LogWriterFactoryInterface* log_writer_factory) {
- if (!log_writer_factory) {
- return std::make_unique<RtcEventLogNull>();
- }
- auto event_log = RtcEventLogFactory(task_queue_factory)
- .CreateRtcEventLog(RtcEventLog::EncodingType::NewFormat);
- bool success = event_log->StartLogging(log_writer_factory->Create(".rtc.dat"),
+ const Environment& env,
+ LogWriterFactoryInterface& log_writer_factory) {
+ auto event_log = RtcEventLogFactory().Create(env);
+ bool success = event_log->StartLogging(log_writer_factory.Create(".rtc.dat"),
kEventLogOutputIntervalMs);
RTC_CHECK(success);
return event_log;
@@ -219,22 +210,25 @@ CallClient::CallClient(
std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
CallClientConfig config)
: time_controller_(time_controller),
- clock_(time_controller->GetClock()),
+ env_(CreateEnvironment(time_controller_->CreateTaskQueueFactory(),
+ time_controller_->GetClock())),
log_writer_factory_(std::move(log_writer_factory)),
network_controller_factory_(log_writer_factory_.get(), config.transport),
- task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
+ task_queue_(env_.task_queue_factory().CreateTaskQueue(
"CallClient",
TaskQueueFactory::Priority::NORMAL)) {
- config.field_trials = &field_trials_;
SendTask([this, config] {
- event_log_ = CreateEventLog(time_controller_->GetTaskQueueFactory(),
- log_writer_factory_.get());
+ if (log_writer_factory_ != nullptr) {
+ EnvironmentFactory env_factory(env_);
+ env_factory.Set(CreateEventLog(env_, *log_writer_factory_));
+ env_ = env_factory.Create();
+ }
fake_audio_setup_ = InitAudio(time_controller_);
- call_ =
- CreateCall(time_controller_, event_log_.get(), config,
- &network_controller_factory_, fake_audio_setup_.audio_state);
- transport_ = std::make_unique<NetworkNodeTransport>(clock_, call_.get());
+ call_ = CreateCall(env_, config, &network_controller_factory_,
+ fake_audio_setup_.audio_state);
+ transport_ =
+ std::make_unique<NetworkNodeTransport>(&env_.clock(), call_.get());
});
}
@@ -243,9 +237,8 @@ CallClient::~CallClient() {
call_.reset();
fake_audio_setup_ = {};
rtc::Event done;
- event_log_->StopLogging([&done] { done.Set(); });
+ env_.event_log().StopLogging([&done] { done.Set(); });
done.Wait(rtc::Event::kForever);
- event_log_.reset();
});
}
@@ -283,7 +276,7 @@ DataRate CallClient::padding_rate() const {
void CallClient::SetRemoteBitrate(DataRate bitrate) {
RemoteBitrateReport msg;
msg.bandwidth = bitrate;
- msg.receive_time = clock_->CurrentTime();
+ msg.receive_time = env_.clock().CurrentTime();
network_controller_factory_.SetRemoteBitrateEstimate(msg);
}
diff --git a/third_party/libwebrtc/test/scenario/call_client.h b/third_party/libwebrtc/test/scenario/call_client.h
index 3717a7e796..f3c483ef28 100644
--- a/third_party/libwebrtc/test/scenario/call_client.h
+++ b/third_party/libwebrtc/test/scenario/call_client.h
@@ -17,6 +17,7 @@
#include <vector>
#include "api/array_view.h"
+#include "api/environment/environment.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_parameters.h"
#include "api/test/time_controller.h"
@@ -156,9 +157,8 @@ class CallClient : public EmulatedNetworkReceiverInterface {
void UnBind();
TimeController* const time_controller_;
- Clock* clock_;
+ Environment env_;
const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_;
- std::unique_ptr<RtcEventLog> event_log_;
LoggingNetworkControllerFactory network_controller_factory_;
CallClientFakeAudio fake_audio_setup_;
std::unique_ptr<Call> call_;
@@ -175,8 +175,6 @@ class CallClient : public EmulatedNetworkReceiverInterface {
std::map<uint32_t, MediaType> ssrc_media_types_;
// Defined last so it's destroyed first.
TaskQueueForTest task_queue_;
-
- const FieldTrialBasedConfig field_trials_;
};
class CallClientPair {
diff --git a/third_party/libwebrtc/test/scenario/scenario_config.h b/third_party/libwebrtc/test/scenario/scenario_config.h
index 9ce99401d7..50845cd677 100644
--- a/third_party/libwebrtc/test/scenario/scenario_config.h
+++ b/third_party/libwebrtc/test/scenario/scenario_config.h
@@ -57,7 +57,6 @@ struct CallClientConfig {
// The number of bites that can be sent in one burst is pacer_burst_interval *
// current bwe. 40ms is the default Chrome setting.
TimeDelta pacer_burst_interval = TimeDelta::Millis(40);
- const FieldTrialsView* field_trials = nullptr;
};
struct PacketStreamConfig {
diff --git a/third_party/libwebrtc/test/scenario/video_stream.cc b/third_party/libwebrtc/test/scenario/video_stream.cc
index 38e42c88e0..eb20f8dbc7 100644
--- a/third_party/libwebrtc/test/scenario/video_stream.cc
+++ b/third_party/libwebrtc/test/scenario/video_stream.cc
@@ -372,9 +372,9 @@ SendVideoStream::SendVideoStream(CallClient* sender,
VideoFrameMatcher* matcher)
: sender_(sender), config_(config) {
video_capturer_ = std::make_unique<FrameGeneratorCapturer>(
- sender_->clock_, CreateFrameGenerator(sender_->clock_, config.source),
- config.source.framerate,
- *sender->time_controller_->GetTaskQueueFactory());
+ &sender_->env_.clock(),
+ CreateFrameGenerator(&sender_->env_.clock(), config.source),
+ config.source.framerate, sender_->env_.task_queue_factory());
video_capturer_->Init();
using Encoder = VideoStreamConfig::Encoder;
@@ -386,9 +386,11 @@ SendVideoStream::SendVideoStream(CallClient* sender,
MutexLock lock(&mutex_);
std::unique_ptr<FakeEncoder> encoder;
if (config_.encoder.codec == Codec::kVideoCodecVP8) {
- encoder = std::make_unique<test::FakeVp8Encoder>(sender_->clock_);
+ encoder = std::make_unique<test::FakeVp8Encoder>(
+ &sender_->env_.clock());
} else if (config_.encoder.codec == Codec::kVideoCodecGeneric) {
- encoder = std::make_unique<test::FakeEncoder>(sender_->clock_);
+ encoder =
+ std::make_unique<test::FakeEncoder>(&sender_->env_.clock());
} else {
RTC_DCHECK_NOTREACHED();
}
@@ -436,7 +438,7 @@ SendVideoStream::SendVideoStream(CallClient* sender,
if (matcher->Active()) {
frame_tap_ = std::make_unique<ForwardingCapturedFrameTap>(
- sender_->clock_, matcher, video_capturer_.get());
+ &sender_->env_.clock(), matcher, video_capturer_.get());
send_stream_->SetSource(frame_tap_.get(),
config.encoder.degradation_preference);
} else {
@@ -565,8 +567,8 @@ ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
for (size_t i = 0; i < num_streams; ++i) {
rtc::VideoSinkInterface<VideoFrame>* renderer = &fake_renderer_;
if (matcher->Active()) {
- render_taps_.emplace_back(
- std::make_unique<DecodedFrameTap>(receiver_->clock_, matcher, i));
+ render_taps_.emplace_back(std::make_unique<DecodedFrameTap>(
+ &receiver_->env_.clock(), matcher, i));
renderer = render_taps_.back().get();
}
auto recv_config = CreateVideoReceiveStreamConfig(
diff --git a/third_party/libwebrtc/test/video_codec_tester.cc b/third_party/libwebrtc/test/video_codec_tester.cc
new file mode 100644
index 0000000000..9453c3a7ef
--- /dev/null
+++ b/third_party/libwebrtc/test/video_codec_tester.cc
@@ -0,0 +1,1324 @@
+/*
+ * 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 "test/video_codec_tester.h"
+
+#include <algorithm>
+#include <set>
+#include <tuple>
+#include <utility>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_bitrate_allocator.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_encoder.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/av1/av1_svc_config.h"
+#include "modules/video_coding/codecs/vp9/svc_config.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+#include "test/testsupport/video_frame_writer.h"
+#include "third_party/libyuv/include/libyuv/compare.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using CodedVideoSource = VideoCodecTester::CodedVideoSource;
+using VideoSourceSettings = VideoCodecTester::VideoSourceSettings;
+using EncodingSettings = VideoCodecTester::EncodingSettings;
+using LayerSettings = EncodingSettings::LayerSettings;
+using LayerId = VideoCodecTester::LayerId;
+using EncoderSettings = VideoCodecTester::EncoderSettings;
+using DecoderSettings = VideoCodecTester::DecoderSettings;
+using PacingSettings = VideoCodecTester::PacingSettings;
+using PacingMode = PacingSettings::PacingMode;
+using VideoCodecStats = VideoCodecTester::VideoCodecStats;
+using DecodeCallback =
+ absl::AnyInvocable<void(const VideoFrame& decoded_frame)>;
+using webrtc::test::ImprovementDirection;
+
+constexpr Frequency k90kHz = Frequency::Hertz(90000);
+
+const std::set<ScalabilityMode> kFullSvcScalabilityModes{
+ ScalabilityMode::kL2T1, ScalabilityMode::kL2T1h, ScalabilityMode::kL2T2,
+ ScalabilityMode::kL2T2h, ScalabilityMode::kL2T3, ScalabilityMode::kL2T3h,
+ ScalabilityMode::kL3T1, ScalabilityMode::kL3T1h, ScalabilityMode::kL3T2,
+ ScalabilityMode::kL3T2h, ScalabilityMode::kL3T3, ScalabilityMode::kL3T3h};
+
+const std::set<ScalabilityMode> kKeySvcScalabilityModes{
+ ScalabilityMode::kL2T1_KEY, ScalabilityMode::kL2T2_KEY,
+ ScalabilityMode::kL2T2_KEY_SHIFT, ScalabilityMode::kL2T3_KEY,
+ ScalabilityMode::kL3T1_KEY, ScalabilityMode::kL3T2_KEY,
+ ScalabilityMode::kL3T3_KEY};
+
+// A thread-safe raw video frame reader.
+class VideoSource {
+ public:
+ explicit VideoSource(VideoSourceSettings source_settings)
+ : source_settings_(source_settings) {
+ MutexLock lock(&mutex_);
+ frame_reader_ = CreateYuvFrameReader(
+ source_settings_.file_path, source_settings_.resolution,
+ YuvFrameReaderImpl::RepeatMode::kPingPong);
+ RTC_CHECK(frame_reader_);
+ }
+
+ // Pulls next frame.
+ VideoFrame PullFrame(uint32_t timestamp_rtp,
+ Resolution resolution,
+ Frequency framerate) {
+ MutexLock lock(&mutex_);
+ int frame_num;
+ auto buffer = frame_reader_->PullFrame(
+ &frame_num, resolution,
+ {.num = framerate.millihertz<int>(),
+ .den = source_settings_.framerate.millihertz<int>()});
+ RTC_CHECK(buffer) << "Can not pull frame. RTP timestamp " << timestamp_rtp;
+ frame_num_[timestamp_rtp] = frame_num;
+ return VideoFrame::Builder()
+ .set_video_frame_buffer(buffer)
+ .set_timestamp_rtp(timestamp_rtp)
+ .set_timestamp_us((timestamp_rtp / k90kHz).us())
+ .build();
+ }
+
+ // Reads frame specified by `timestamp_rtp`, scales it to `resolution` and
+ // returns. Frame with the given `timestamp_rtp` is expected to be pulled
+ // before.
+ VideoFrame ReadFrame(uint32_t timestamp_rtp, Resolution resolution) {
+ MutexLock lock(&mutex_);
+ RTC_CHECK(frame_num_.find(timestamp_rtp) != frame_num_.end())
+ << "Frame with RTP timestamp " << timestamp_rtp
+ << " was not pulled before";
+ auto buffer =
+ frame_reader_->ReadFrame(frame_num_.at(timestamp_rtp), resolution);
+ return VideoFrame::Builder()
+ .set_video_frame_buffer(buffer)
+ .set_timestamp_rtp(timestamp_rtp)
+ .build();
+ }
+
+ private:
+ VideoSourceSettings source_settings_;
+ std::unique_ptr<FrameReader> frame_reader_ RTC_GUARDED_BY(mutex_);
+ std::map<uint32_t, int> frame_num_ RTC_GUARDED_BY(mutex_);
+ Mutex mutex_;
+};
+
+// Pacer calculates delay necessary to keep frame encode or decode call spaced
+// from the previous calls by the pacing time. `Schedule` is expected to be
+// called as close as possible to posting frame encode or decode task. This
+// class is not thread safe.
+class Pacer {
+ public:
+ explicit Pacer(PacingSettings settings)
+ : settings_(settings), delay_(TimeDelta::Zero()) {}
+
+ Timestamp Schedule(Timestamp timestamp) {
+ Timestamp now = Timestamp::Micros(rtc::TimeMicros());
+ if (settings_.mode == PacingMode::kNoPacing) {
+ return now;
+ }
+
+ Timestamp scheduled = now;
+ if (prev_scheduled_) {
+ scheduled = *prev_scheduled_ + PacingTime(timestamp);
+ if (scheduled < now) {
+ scheduled = now;
+ }
+ }
+
+ prev_timestamp_ = timestamp;
+ prev_scheduled_ = scheduled;
+ return scheduled;
+ }
+
+ private:
+ TimeDelta PacingTime(Timestamp timestamp) {
+ if (settings_.mode == PacingMode::kRealTime) {
+ return timestamp - *prev_timestamp_;
+ }
+ RTC_CHECK_EQ(PacingMode::kConstantRate, settings_.mode);
+ return 1 / settings_.constant_rate;
+ }
+
+ PacingSettings settings_;
+ absl::optional<Timestamp> prev_timestamp_;
+ absl::optional<Timestamp> prev_scheduled_;
+ TimeDelta delay_;
+};
+
+class LimitedTaskQueue {
+ public:
+ // The codec tester reads frames from video source in the main thread.
+ // Encoding and decoding are done in separate threads. If encoding or
+ // decoding is slow, the reading may go far ahead and may buffer too many
+ // frames in memory. To prevent this we limit the encoding/decoding queue
+ // size. When the queue is full, the main thread and, hence, reading frames
+ // from video source is blocked until a previously posted encoding/decoding
+ // task starts.
+ static constexpr int kMaxTaskQueueSize = 3;
+
+ LimitedTaskQueue() : queue_size_(0) {}
+
+ void PostScheduledTask(absl::AnyInvocable<void() &&> task, Timestamp start) {
+ ++queue_size_;
+ task_queue_.PostTask([this, task = std::move(task), start]() mutable {
+ // `TaskQueue` doesn't guarantee FIFO order of execution for delayed
+ // tasks.
+ int64_t wait_ms = (start - Timestamp::Millis(rtc::TimeMillis())).ms();
+ if (wait_ms > 0) {
+ RTC_CHECK_LT(wait_ms, 10000) << "Too high wait_ms " << wait_ms;
+ SleepMs(wait_ms);
+ }
+ std::move(task)();
+ --queue_size_;
+ task_executed_.Set();
+ });
+
+ task_executed_.Reset();
+ if (queue_size_ > kMaxTaskQueueSize) {
+ task_executed_.Wait(rtc::Event::kForever);
+ RTC_CHECK(queue_size_ <= kMaxTaskQueueSize);
+ }
+ }
+
+ void PostTaskAndWait(absl::AnyInvocable<void() &&> task) {
+ PostScheduledTask(std::move(task), Timestamp::Millis(rtc::TimeMillis()));
+ task_queue_.WaitForPreviouslyPostedTasks();
+ }
+
+ private:
+ TaskQueueForTest task_queue_;
+ std::atomic_int queue_size_;
+ rtc::Event task_executed_;
+};
+
+class TesterY4mWriter {
+ public:
+ explicit TesterY4mWriter(absl::string_view base_path)
+ : base_path_(base_path) {}
+
+ ~TesterY4mWriter() {
+ task_queue_.SendTask([] {});
+ }
+
+ void Write(const VideoFrame& frame, int spatial_idx) {
+ task_queue_.PostTask([this, frame, spatial_idx] {
+ if (y4m_writers_.find(spatial_idx) == y4m_writers_.end()) {
+ std::string file_path =
+ base_path_ + "-s" + std::to_string(spatial_idx) + ".y4m";
+ Y4mVideoFrameWriterImpl* y4m_writer = new Y4mVideoFrameWriterImpl(
+ file_path, frame.width(), frame.height(), /*fps=*/30);
+ RTC_CHECK(y4m_writer);
+
+ y4m_writers_[spatial_idx] =
+ std::unique_ptr<VideoFrameWriter>(y4m_writer);
+ }
+
+ y4m_writers_.at(spatial_idx)->WriteFrame(frame);
+ });
+ }
+
+ private:
+ std::string base_path_;
+ std::map<int, std::unique_ptr<VideoFrameWriter>> y4m_writers_;
+ TaskQueueForTest task_queue_;
+};
+
+class TesterIvfWriter {
+ public:
+ explicit TesterIvfWriter(absl::string_view base_path)
+ : base_path_(base_path) {}
+
+ ~TesterIvfWriter() {
+ task_queue_.SendTask([] {});
+ }
+
+ void Write(const EncodedImage& encoded_frame) {
+ task_queue_.PostTask([this, encoded_frame] {
+ int spatial_idx = encoded_frame.SimulcastIndex().value_or(0);
+ if (ivf_file_writers_.find(spatial_idx) == ivf_file_writers_.end()) {
+ std::string ivf_path =
+ base_path_ + "-s" + std::to_string(spatial_idx) + ".ivf";
+ FileWrapper ivf_file = FileWrapper::OpenWriteOnly(ivf_path);
+ RTC_CHECK(ivf_file.is_open());
+
+ std::unique_ptr<IvfFileWriter> ivf_writer =
+ IvfFileWriter::Wrap(std::move(ivf_file), /*byte_limit=*/0);
+ RTC_CHECK(ivf_writer);
+
+ ivf_file_writers_[spatial_idx] = std::move(ivf_writer);
+ }
+
+ // To play: ffplay -vcodec vp8|vp9|av1|hevc|h264 filename
+ ivf_file_writers_.at(spatial_idx)
+ ->WriteFrame(encoded_frame, VideoCodecType::kVideoCodecGeneric);
+ });
+ }
+
+ private:
+ std::string base_path_;
+ std::map<int, std::unique_ptr<IvfFileWriter>> ivf_file_writers_;
+ TaskQueueForTest task_queue_;
+};
+
+class LeakyBucket {
+ public:
+ LeakyBucket() : level_bits_(0) {}
+
+ // Updates bucket level and returns its current level in bits. Data is remove
+ // from bucket with rate equal to target bitrate of previous frame. Bucket
+ // level is tracked with floating point precision. Returned value of bucket
+ // level is rounded up.
+ int Update(const VideoCodecStats::Frame& frame) {
+ RTC_CHECK(frame.target_bitrate) << "Bitrate must be specified.";
+ if (prev_frame_) {
+ RTC_CHECK_GT(frame.timestamp_rtp, prev_frame_->timestamp_rtp)
+ << "Timestamp must increase.";
+ TimeDelta passed =
+ (frame.timestamp_rtp - prev_frame_->timestamp_rtp) / k90kHz;
+ level_bits_ -=
+ prev_frame_->target_bitrate->bps<double>() * passed.seconds<double>();
+ level_bits_ = std::max(level_bits_, 0.0);
+ }
+ prev_frame_ = frame;
+ level_bits_ += frame.frame_size.bytes() * 8;
+ return static_cast<int>(std::ceil(level_bits_));
+ }
+
+ private:
+ absl::optional<VideoCodecStats::Frame> prev_frame_;
+ double level_bits_;
+};
+
+class VideoCodecAnalyzer : public VideoCodecTester::VideoCodecStats {
+ public:
+ explicit VideoCodecAnalyzer(VideoSource* video_source)
+ : video_source_(video_source) {}
+
+ void StartEncode(const VideoFrame& video_frame,
+ const EncodingSettings& encoding_settings) {
+ int64_t encode_start_us = rtc::TimeMicros();
+ task_queue_.PostTask([this, timestamp_rtp = video_frame.timestamp(),
+ encoding_settings, encode_start_us]() {
+ RTC_CHECK(frames_.find(timestamp_rtp) == frames_.end())
+ << "Duplicate frame. Frame with timestamp " << timestamp_rtp
+ << " was seen before";
+
+ Frame frame;
+ frame.timestamp_rtp = timestamp_rtp;
+ frame.encode_start = Timestamp::Micros(encode_start_us),
+ frames_.emplace(timestamp_rtp,
+ std::map<int, Frame>{{/*spatial_idx=*/0, frame}});
+ encoding_settings_.emplace(timestamp_rtp, encoding_settings);
+ });
+ }
+
+ void FinishEncode(const EncodedImage& encoded_frame) {
+ int64_t encode_finished_us = rtc::TimeMicros();
+ task_queue_.PostTask(
+ [this, timestamp_rtp = encoded_frame.RtpTimestamp(),
+ spatial_idx = encoded_frame.SpatialIndex().value_or(0),
+ temporal_idx = encoded_frame.TemporalIndex().value_or(0),
+ width = encoded_frame._encodedWidth,
+ height = encoded_frame._encodedHeight,
+ frame_type = encoded_frame._frameType,
+ frame_size_bytes = encoded_frame.size(), qp = encoded_frame.qp_,
+ encode_finished_us]() {
+ if (spatial_idx > 0) {
+ RTC_CHECK(frames_.find(timestamp_rtp) != frames_.end())
+ << "Spatial layer 0 frame with timestamp " << timestamp_rtp
+ << " was not seen before";
+ const Frame& base_frame =
+ frames_.at(timestamp_rtp).at(/*spatial_idx=*/0);
+ frames_.at(timestamp_rtp).emplace(spatial_idx, base_frame);
+ }
+
+ Frame& frame = frames_.at(timestamp_rtp).at(spatial_idx);
+ frame.layer_id = {.spatial_idx = spatial_idx,
+ .temporal_idx = temporal_idx};
+ frame.width = width;
+ frame.height = height;
+ frame.frame_size = DataSize::Bytes(frame_size_bytes);
+ frame.qp = qp;
+ frame.keyframe = frame_type == VideoFrameType::kVideoFrameKey;
+ frame.encode_time =
+ Timestamp::Micros(encode_finished_us) - frame.encode_start;
+ frame.encoded = true;
+ });
+ }
+
+ void StartDecode(const EncodedImage& encoded_frame) {
+ int64_t decode_start_us = rtc::TimeMicros();
+ task_queue_.PostTask(
+ [this, timestamp_rtp = encoded_frame.RtpTimestamp(),
+ spatial_idx = encoded_frame.SpatialIndex().value_or(0),
+ frame_size_bytes = encoded_frame.size(), decode_start_us]() {
+ if (frames_.find(timestamp_rtp) == frames_.end() ||
+ frames_.at(timestamp_rtp).find(spatial_idx) ==
+ frames_.at(timestamp_rtp).end()) {
+ Frame frame;
+ frame.timestamp_rtp = timestamp_rtp;
+ frame.layer_id = {.spatial_idx = spatial_idx};
+ frame.frame_size = DataSize::Bytes(frame_size_bytes);
+ frames_.emplace(timestamp_rtp,
+ std::map<int, Frame>{{spatial_idx, frame}});
+ }
+
+ Frame& frame = frames_.at(timestamp_rtp).at(spatial_idx);
+ frame.decode_start = Timestamp::Micros(decode_start_us);
+ });
+ }
+
+ void FinishDecode(const VideoFrame& decoded_frame, int spatial_idx) {
+ int64_t decode_finished_us = rtc::TimeMicros();
+ task_queue_.PostTask([this, timestamp_rtp = decoded_frame.timestamp(),
+ spatial_idx, width = decoded_frame.width(),
+ height = decoded_frame.height(),
+ decode_finished_us]() {
+ Frame& frame = frames_.at(timestamp_rtp).at(spatial_idx);
+ frame.decode_time =
+ Timestamp::Micros(decode_finished_us) - frame.decode_start;
+ if (!frame.encoded) {
+ frame.width = width;
+ frame.height = height;
+ }
+ frame.decoded = true;
+ });
+
+ if (video_source_ != nullptr) {
+ // Copy hardware-backed frame into main memory to release output buffers
+ // which number may be limited in hardware decoders.
+ rtc::scoped_refptr<I420BufferInterface> decoded_buffer =
+ decoded_frame.video_frame_buffer()->ToI420();
+
+ task_queue_.PostTask([this, decoded_buffer,
+ timestamp_rtp = decoded_frame.timestamp(),
+ spatial_idx]() {
+ VideoFrame ref_frame = video_source_->ReadFrame(
+ timestamp_rtp, {.width = decoded_buffer->width(),
+ .height = decoded_buffer->height()});
+ rtc::scoped_refptr<I420BufferInterface> ref_buffer =
+ ref_frame.video_frame_buffer()->ToI420();
+ Frame& frame = frames_.at(timestamp_rtp).at(spatial_idx);
+ frame.psnr = CalcPsnr(*decoded_buffer, *ref_buffer);
+ });
+ }
+ }
+
+ std::vector<Frame> Slice(Filter filter, bool merge) const {
+ std::vector<Frame> slice;
+ for (const auto& [timestamp_rtp, temporal_unit_frames] : frames_) {
+ if (temporal_unit_frames.empty()) {
+ continue;
+ }
+
+ bool is_svc = false;
+ if (!encoding_settings_.empty()) {
+ ScalabilityMode scalability_mode =
+ encoding_settings_.at(timestamp_rtp).scalability_mode;
+ if (kFullSvcScalabilityModes.count(scalability_mode) > 0 ||
+ (kKeySvcScalabilityModes.count(scalability_mode) > 0 &&
+ temporal_unit_frames.at(0).keyframe)) {
+ is_svc = true;
+ }
+ }
+
+ std::vector<Frame> subframes;
+ for (const auto& [spatial_idx, frame] : temporal_unit_frames) {
+ if (frame.timestamp_rtp < filter.min_timestamp_rtp ||
+ frame.timestamp_rtp > filter.max_timestamp_rtp) {
+ continue;
+ }
+ if (filter.layer_id) {
+ if (is_svc &&
+ frame.layer_id.spatial_idx > filter.layer_id->spatial_idx) {
+ continue;
+ }
+ if (!is_svc &&
+ frame.layer_id.spatial_idx != filter.layer_id->spatial_idx) {
+ continue;
+ }
+ if (frame.layer_id.temporal_idx > filter.layer_id->temporal_idx) {
+ continue;
+ }
+ }
+ subframes.push_back(frame);
+ }
+
+ if (subframes.empty()) {
+ continue;
+ }
+
+ if (!merge) {
+ std::copy(subframes.begin(), subframes.end(),
+ std::back_inserter(slice));
+ continue;
+ }
+
+ Frame superframe = subframes.back();
+ for (const Frame& frame :
+ rtc::ArrayView<Frame>(subframes).subview(0, subframes.size() - 1)) {
+ superframe.frame_size += frame.frame_size;
+ superframe.keyframe |= frame.keyframe;
+ superframe.encode_time =
+ std::max(superframe.encode_time, frame.encode_time);
+ superframe.decode_time =
+ std::max(superframe.decode_time, frame.decode_time);
+ }
+
+ if (!encoding_settings_.empty()) {
+ RTC_CHECK(encoding_settings_.find(superframe.timestamp_rtp) !=
+ encoding_settings_.end())
+ << "No encoding settings for frame " << superframe.timestamp_rtp;
+ const EncodingSettings& es =
+ encoding_settings_.at(superframe.timestamp_rtp);
+ superframe.target_bitrate = GetTargetBitrate(es, filter.layer_id);
+ superframe.target_framerate = GetTargetFramerate(es, filter.layer_id);
+ }
+
+ slice.push_back(superframe);
+ }
+ return slice;
+ }
+
+ Stream Aggregate(Filter filter) const {
+ std::vector<Frame> frames = Slice(filter, /*merge=*/true);
+ Stream stream;
+ LeakyBucket leaky_bucket;
+ for (const Frame& frame : frames) {
+ Timestamp time = Timestamp::Micros((frame.timestamp_rtp / k90kHz).us());
+ if (!frame.frame_size.IsZero()) {
+ stream.width.AddSample(StatsSample(frame.width, time));
+ stream.height.AddSample(StatsSample(frame.height, time));
+ stream.frame_size_bytes.AddSample(
+ StatsSample(frame.frame_size.bytes(), time));
+ stream.keyframe.AddSample(StatsSample(frame.keyframe, time));
+ if (frame.qp) {
+ stream.qp.AddSample(StatsSample(*frame.qp, time));
+ }
+ }
+ if (frame.encoded) {
+ stream.encode_time_ms.AddSample(
+ StatsSample(frame.encode_time.ms(), time));
+ }
+ if (frame.decoded) {
+ stream.decode_time_ms.AddSample(
+ StatsSample(frame.decode_time.ms(), time));
+ }
+ if (frame.psnr) {
+ stream.psnr.y.AddSample(StatsSample(frame.psnr->y, time));
+ stream.psnr.u.AddSample(StatsSample(frame.psnr->u, time));
+ stream.psnr.v.AddSample(StatsSample(frame.psnr->v, time));
+ }
+ if (frame.target_framerate) {
+ stream.target_framerate_fps.AddSample(
+ StatsSample(frame.target_framerate->hertz<double>(), time));
+ }
+ if (frame.target_bitrate) {
+ stream.target_bitrate_kbps.AddSample(
+ StatsSample(frame.target_bitrate->kbps<double>(), time));
+ int buffer_level_bits = leaky_bucket.Update(frame);
+ stream.transmission_time_ms.AddSample(StatsSample(
+ 1000 * buffer_level_bits / frame.target_bitrate->bps<double>(),
+ time));
+ }
+ }
+
+ int num_encoded_frames = stream.frame_size_bytes.NumSamples();
+ const Frame& first_frame = frames.front();
+
+ Filter filter_all_layers{.min_timestamp_rtp = filter.min_timestamp_rtp,
+ .max_timestamp_rtp = filter.max_timestamp_rtp};
+ std::vector<Frame> frames_all_layers =
+ Slice(filter_all_layers, /*merge=*/true);
+ const Frame& last_frame = frames_all_layers.back();
+ TimeDelta duration =
+ (last_frame.timestamp_rtp - first_frame.timestamp_rtp) / k90kHz;
+ if (last_frame.target_framerate) {
+ duration += 1 / *last_frame.target_framerate;
+ }
+
+ DataRate encoded_bitrate =
+ DataSize::Bytes(stream.frame_size_bytes.GetSum()) / duration;
+ Frequency encoded_framerate = num_encoded_frames / duration;
+
+ double bitrate_mismatch_pct = 0.0;
+ if (const auto& target_bitrate = first_frame.target_bitrate;
+ target_bitrate) {
+ bitrate_mismatch_pct = 100 * (encoded_bitrate / *target_bitrate - 1);
+ }
+ double framerate_mismatch_pct = 0.0;
+ if (const auto& target_framerate = first_frame.target_framerate;
+ target_framerate) {
+ framerate_mismatch_pct =
+ 100 * (encoded_framerate / *target_framerate - 1);
+ }
+
+ for (Frame& frame : frames) {
+ Timestamp time = Timestamp::Micros((frame.timestamp_rtp / k90kHz).us());
+ stream.encoded_bitrate_kbps.AddSample(
+ StatsSample(encoded_bitrate.kbps<double>(), time));
+ stream.encoded_framerate_fps.AddSample(
+ StatsSample(encoded_framerate.hertz<double>(), time));
+ stream.bitrate_mismatch_pct.AddSample(
+ StatsSample(bitrate_mismatch_pct, time));
+ stream.framerate_mismatch_pct.AddSample(
+ StatsSample(framerate_mismatch_pct, time));
+ }
+
+ return stream;
+ }
+
+ void LogMetrics(absl::string_view csv_path,
+ std::vector<Frame> frames,
+ std::map<std::string, std::string> metadata) const {
+ RTC_LOG(LS_INFO) << "Write metrics to " << csv_path;
+ FILE* csv_file = fopen(csv_path.data(), "w");
+ const std::string delimiter = ";";
+ rtc::StringBuilder header;
+ header
+ << "timestamp_rtp;spatial_idx;temporal_idx;width;height;frame_size_"
+ "bytes;keyframe;qp;encode_time_us;decode_time_us;psnr_y_db;psnr_u_"
+ "db;psnr_v_db;target_bitrate_kbps;target_framerate_fps";
+ for (const auto& data : metadata) {
+ header << ";" << data.first;
+ }
+ fwrite(header.str().c_str(), 1, header.size(), csv_file);
+
+ for (const Frame& f : frames) {
+ rtc::StringBuilder row;
+ row << "\n" << f.timestamp_rtp;
+ row << ";" << f.layer_id.spatial_idx;
+ row << ";" << f.layer_id.temporal_idx;
+ row << ";" << f.width;
+ row << ";" << f.height;
+ row << ";" << f.frame_size.bytes();
+ row << ";" << f.keyframe;
+ row << ";";
+ if (f.qp) {
+ row << *f.qp;
+ }
+ row << ";" << f.encode_time.us();
+ row << ";" << f.decode_time.us();
+ if (f.psnr) {
+ row << ";" << f.psnr->y;
+ row << ";" << f.psnr->u;
+ row << ";" << f.psnr->v;
+ } else {
+ row << ";;;";
+ }
+
+ const auto& es = encoding_settings_.at(f.timestamp_rtp);
+ row << ";"
+ << f.target_bitrate.value_or(GetTargetBitrate(es, f.layer_id)).kbps();
+ row << ";"
+ << f.target_framerate.value_or(GetTargetFramerate(es, f.layer_id))
+ .hertz<double>();
+
+ for (const auto& data : metadata) {
+ row << ";" << data.second;
+ }
+ fwrite(row.str().c_str(), 1, row.size(), csv_file);
+ }
+
+ fclose(csv_file);
+ }
+
+ void Flush() { task_queue_.WaitForPreviouslyPostedTasks(); }
+
+ private:
+ struct FrameId {
+ uint32_t timestamp_rtp;
+ int spatial_idx;
+
+ bool operator==(const FrameId& o) const {
+ return timestamp_rtp == o.timestamp_rtp && spatial_idx == o.spatial_idx;
+ }
+ bool operator<(const FrameId& o) const {
+ return timestamp_rtp < o.timestamp_rtp ||
+ (timestamp_rtp == o.timestamp_rtp && spatial_idx < o.spatial_idx);
+ }
+ };
+
+ Frame::Psnr CalcPsnr(const I420BufferInterface& ref_buffer,
+ const I420BufferInterface& dec_buffer) {
+ RTC_CHECK_EQ(ref_buffer.width(), dec_buffer.width());
+ RTC_CHECK_EQ(ref_buffer.height(), dec_buffer.height());
+
+ uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
+ dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(),
+ ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height());
+
+ uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
+ dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(),
+ ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+ uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
+ dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(),
+ ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2);
+
+ int num_y_samples = dec_buffer.width() * dec_buffer.height();
+ Frame::Psnr psnr;
+ psnr.y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples);
+ psnr.u = libyuv::SumSquareErrorToPsnr(sse_u, num_y_samples / 4);
+ psnr.v = libyuv::SumSquareErrorToPsnr(sse_v, num_y_samples / 4);
+ return psnr;
+ }
+
+ DataRate GetTargetBitrate(const EncodingSettings& encoding_settings,
+ absl::optional<LayerId> layer_id) const {
+ int base_spatial_idx;
+ if (layer_id.has_value()) {
+ bool is_svc =
+ kFullSvcScalabilityModes.count(encoding_settings.scalability_mode);
+ base_spatial_idx = is_svc ? 0 : layer_id->spatial_idx;
+ } else {
+ int num_spatial_layers =
+ ScalabilityModeToNumSpatialLayers(encoding_settings.scalability_mode);
+ int num_temporal_layers = ScalabilityModeToNumTemporalLayers(
+ encoding_settings.scalability_mode);
+ layer_id = LayerId({.spatial_idx = num_spatial_layers - 1,
+ .temporal_idx = num_temporal_layers - 1});
+ base_spatial_idx = 0;
+ }
+
+ DataRate bitrate = DataRate::Zero();
+ for (int sidx = base_spatial_idx; sidx <= layer_id->spatial_idx; ++sidx) {
+ for (int tidx = 0; tidx <= layer_id->temporal_idx; ++tidx) {
+ auto layer_settings = encoding_settings.layers_settings.find(
+ {.spatial_idx = sidx, .temporal_idx = tidx});
+ RTC_CHECK(layer_settings != encoding_settings.layers_settings.end())
+ << "bitrate is not specified for layer sidx=" << sidx
+ << " tidx=" << tidx;
+ bitrate += layer_settings->second.bitrate;
+ }
+ }
+ return bitrate;
+ }
+
+ Frequency GetTargetFramerate(const EncodingSettings& encoding_settings,
+ absl::optional<LayerId> layer_id) const {
+ if (layer_id.has_value()) {
+ auto layer_settings = encoding_settings.layers_settings.find(
+ {.spatial_idx = layer_id->spatial_idx,
+ .temporal_idx = layer_id->temporal_idx});
+ RTC_CHECK(layer_settings != encoding_settings.layers_settings.end())
+ << "framerate is not specified for layer sidx="
+ << layer_id->spatial_idx << " tidx=" << layer_id->temporal_idx;
+ return layer_settings->second.framerate;
+ }
+ return encoding_settings.layers_settings.rbegin()->second.framerate;
+ }
+
+ SamplesStatsCounter::StatsSample StatsSample(double value,
+ Timestamp time) const {
+ return SamplesStatsCounter::StatsSample{value, time};
+ }
+
+ VideoSource* const video_source_;
+ TaskQueueForTest task_queue_;
+ // RTP timestamp -> spatial layer -> Frame
+ std::map<uint32_t, std::map<int, Frame>> frames_;
+ std::map<uint32_t, EncodingSettings> encoding_settings_;
+};
+
+class Decoder : public DecodedImageCallback {
+ public:
+ Decoder(VideoDecoderFactory* decoder_factory,
+ const DecoderSettings& decoder_settings,
+ VideoCodecAnalyzer* analyzer)
+ : decoder_factory_(decoder_factory),
+ analyzer_(analyzer),
+ pacer_(decoder_settings.pacing_settings) {
+ RTC_CHECK(analyzer_) << "Analyzer must be provided";
+
+ if (decoder_settings.decoder_input_base_path) {
+ ivf_writer_ = std::make_unique<TesterIvfWriter>(
+ *decoder_settings.decoder_input_base_path);
+ }
+
+ if (decoder_settings.decoder_output_base_path) {
+ y4m_writer_ = std::make_unique<TesterY4mWriter>(
+ *decoder_settings.decoder_output_base_path);
+ }
+ }
+
+ void Initialize(const SdpVideoFormat& sdp_video_format) {
+ decoder_ = decoder_factory_->CreateVideoDecoder(sdp_video_format);
+ RTC_CHECK(decoder_) << "Could not create decoder for video format "
+ << sdp_video_format.ToString();
+
+ task_queue_.PostTaskAndWait([this, &sdp_video_format] {
+ decoder_->RegisterDecodeCompleteCallback(this);
+
+ VideoDecoder::Settings ds;
+ ds.set_codec_type(PayloadStringToCodecType(sdp_video_format.name));
+ ds.set_number_of_cores(1);
+ ds.set_max_render_resolution({1280, 720});
+ bool result = decoder_->Configure(ds);
+ RTC_CHECK(result) << "Failed to configure decoder";
+ });
+ }
+
+ void Decode(const EncodedImage& encoded_frame) {
+ Timestamp pts =
+ Timestamp::Micros((encoded_frame.RtpTimestamp() / k90kHz).us());
+
+ task_queue_.PostScheduledTask(
+ [this, encoded_frame] {
+ analyzer_->StartDecode(encoded_frame);
+ int error = decoder_->Decode(encoded_frame, /*render_time_ms*/ 0);
+ if (error != 0) {
+ RTC_LOG(LS_WARNING)
+ << "Decode failed with error code " << error
+ << " RTP timestamp " << encoded_frame.RtpTimestamp();
+ }
+ },
+ pacer_.Schedule(pts));
+
+ if (ivf_writer_) {
+ ivf_writer_->Write(encoded_frame);
+ }
+ }
+
+ void Flush() {
+ // TODO(webrtc:14852): Add Flush() to VideoDecoder API.
+ task_queue_.PostTaskAndWait([this] { decoder_->Release(); });
+ }
+
+ private:
+ int Decoded(VideoFrame& decoded_frame) override {
+ analyzer_->FinishDecode(decoded_frame, /*spatial_idx=*/0);
+
+ if (y4m_writer_) {
+ y4m_writer_->Write(decoded_frame, /*spatial_idx=*/0);
+ }
+
+ return WEBRTC_VIDEO_CODEC_OK;
+ }
+
+ VideoDecoderFactory* decoder_factory_;
+ std::unique_ptr<VideoDecoder> decoder_;
+ VideoCodecAnalyzer* const analyzer_;
+ Pacer pacer_;
+ LimitedTaskQueue task_queue_;
+ std::unique_ptr<TesterIvfWriter> ivf_writer_;
+ std::unique_ptr<TesterY4mWriter> y4m_writer_;
+};
+
+class Encoder : public EncodedImageCallback {
+ public:
+ using EncodeCallback =
+ absl::AnyInvocable<void(const EncodedImage& encoded_frame)>;
+
+ Encoder(VideoEncoderFactory* encoder_factory,
+ const EncoderSettings& encoder_settings,
+ VideoCodecAnalyzer* analyzer)
+ : encoder_factory_(encoder_factory),
+ analyzer_(analyzer),
+ pacer_(encoder_settings.pacing_settings) {
+ RTC_CHECK(analyzer_) << "Analyzer must be provided";
+
+ if (encoder_settings.encoder_input_base_path) {
+ y4m_writer_ = std::make_unique<TesterY4mWriter>(
+ *encoder_settings.encoder_input_base_path);
+ }
+
+ if (encoder_settings.encoder_output_base_path) {
+ ivf_writer_ = std::make_unique<TesterIvfWriter>(
+ *encoder_settings.encoder_output_base_path);
+ }
+ }
+
+ void Initialize(const EncodingSettings& encoding_settings) {
+ encoder_ = encoder_factory_->CreateVideoEncoder(
+ encoding_settings.sdp_video_format);
+ RTC_CHECK(encoder_) << "Could not create encoder for video format "
+ << encoding_settings.sdp_video_format.ToString();
+
+ task_queue_.PostTaskAndWait([this, encoding_settings] {
+ encoder_->RegisterEncodeCompleteCallback(this);
+ Configure(encoding_settings);
+ SetRates(encoding_settings);
+ });
+ }
+
+ void Encode(const VideoFrame& input_frame,
+ const EncodingSettings& encoding_settings,
+ EncodeCallback callback) {
+ {
+ MutexLock lock(&mutex_);
+ callbacks_[input_frame.timestamp()] = std::move(callback);
+ }
+
+ Timestamp pts = Timestamp::Micros((input_frame.timestamp() / k90kHz).us());
+
+ task_queue_.PostScheduledTask(
+ [this, input_frame, encoding_settings] {
+ analyzer_->StartEncode(input_frame, encoding_settings);
+
+ if (!last_encoding_settings_ ||
+ !IsSameRate(encoding_settings, *last_encoding_settings_)) {
+ SetRates(encoding_settings);
+ }
+
+ int error = encoder_->Encode(input_frame, /*frame_types=*/nullptr);
+ if (error != 0) {
+ RTC_LOG(LS_WARNING) << "Encode failed with error code " << error
+ << " RTP timestamp " << input_frame.timestamp();
+ }
+
+ last_encoding_settings_ = encoding_settings;
+ },
+ pacer_.Schedule(pts));
+
+ if (y4m_writer_) {
+ y4m_writer_->Write(input_frame, /*spatial_idx=*/0);
+ }
+ }
+
+ void Flush() {
+ task_queue_.PostTaskAndWait([this] { encoder_->Release(); });
+ }
+
+ private:
+ Result OnEncodedImage(const EncodedImage& encoded_frame,
+ const CodecSpecificInfo* codec_specific_info) override {
+ analyzer_->FinishEncode(encoded_frame);
+
+ {
+ MutexLock lock(&mutex_);
+ auto it = callbacks_.find(encoded_frame.RtpTimestamp());
+ RTC_CHECK(it != callbacks_.end());
+ it->second(encoded_frame);
+ callbacks_.erase(callbacks_.begin(), it);
+ }
+
+ if (ivf_writer_ != nullptr) {
+ ivf_writer_->Write(encoded_frame);
+ }
+
+ return Result(Result::Error::OK);
+ }
+
+ void Configure(const EncodingSettings& es) {
+ const LayerSettings& layer_settings = es.layers_settings.rbegin()->second;
+ const DataRate& bitrate = layer_settings.bitrate;
+
+ VideoCodec vc;
+ vc.width = layer_settings.resolution.width;
+ vc.height = layer_settings.resolution.height;
+ vc.startBitrate = bitrate.kbps();
+ vc.maxBitrate = bitrate.kbps();
+ vc.minBitrate = 0;
+ vc.maxFramerate = layer_settings.framerate.hertz<uint32_t>();
+ vc.active = true;
+ vc.numberOfSimulcastStreams = 0;
+ vc.mode = webrtc::VideoCodecMode::kRealtimeVideo;
+ vc.SetFrameDropEnabled(true);
+ vc.SetScalabilityMode(es.scalability_mode);
+ vc.SetVideoEncoderComplexity(VideoCodecComplexity::kComplexityNormal);
+
+ vc.codecType = PayloadStringToCodecType(es.sdp_video_format.name);
+ switch (vc.codecType) {
+ case kVideoCodecVP8:
+ *(vc.VP8()) = VideoEncoder::GetDefaultVp8Settings();
+ vc.VP8()->SetNumberOfTemporalLayers(
+ ScalabilityModeToNumTemporalLayers(es.scalability_mode));
+ vc.qpMax = cricket::kDefaultVideoMaxQpVpx;
+ // TODO(webrtc:14852): Configure simulcast.
+ break;
+ case kVideoCodecVP9:
+ *(vc.VP9()) = VideoEncoder::GetDefaultVp9Settings();
+ // See LibvpxVp9Encoder::ExplicitlyConfiguredSpatialLayers.
+ vc.spatialLayers[0].targetBitrate = vc.maxBitrate;
+ vc.qpMax = cricket::kDefaultVideoMaxQpVpx;
+ break;
+ case kVideoCodecAV1:
+ vc.qpMax = cricket::kDefaultVideoMaxQpVpx;
+ break;
+ case kVideoCodecH264:
+ *(vc.H264()) = VideoEncoder::GetDefaultH264Settings();
+ vc.qpMax = cricket::kDefaultVideoMaxQpH26x;
+ break;
+ case kVideoCodecH265:
+ vc.qpMax = cricket::kDefaultVideoMaxQpH26x;
+ break;
+ case kVideoCodecGeneric:
+ case kVideoCodecMultiplex:
+ RTC_CHECK_NOTREACHED();
+ break;
+ }
+
+ VideoEncoder::Settings ves(
+ VideoEncoder::Capabilities(/*loss_notification=*/false),
+ /*number_of_cores=*/1,
+ /*max_payload_size=*/1440);
+
+ int result = encoder_->InitEncode(&vc, ves);
+ RTC_CHECK(result == WEBRTC_VIDEO_CODEC_OK);
+
+ SetRates(es);
+ }
+
+ void SetRates(const EncodingSettings& es) {
+ VideoEncoder::RateControlParameters rc;
+ int num_spatial_layers =
+ ScalabilityModeToNumSpatialLayers(es.scalability_mode);
+ int num_temporal_layers =
+ ScalabilityModeToNumTemporalLayers(es.scalability_mode);
+ for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+ for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
+ auto layers_settings = es.layers_settings.find(
+ {.spatial_idx = sidx, .temporal_idx = tidx});
+ RTC_CHECK(layers_settings != es.layers_settings.end())
+ << "Bitrate for layer S=" << sidx << " T=" << tidx << " is not set";
+ rc.bitrate.SetBitrate(sidx, tidx,
+ layers_settings->second.bitrate.bps());
+ }
+ }
+ rc.framerate_fps =
+ es.layers_settings.rbegin()->second.framerate.hertz<double>();
+ encoder_->SetRates(rc);
+ }
+
+ bool IsSameRate(const EncodingSettings& a, const EncodingSettings& b) const {
+ for (auto [layer_id, layer] : a.layers_settings) {
+ const auto& other_layer = b.layers_settings.at(layer_id);
+ if (layer.bitrate != other_layer.bitrate ||
+ layer.framerate != other_layer.framerate) {
+ return false;
+ }
+ }
+
+ return true;
+ }
+
+ VideoEncoderFactory* const encoder_factory_;
+ std::unique_ptr<VideoEncoder> encoder_;
+ VideoCodecAnalyzer* const analyzer_;
+ Pacer pacer_;
+ absl::optional<EncodingSettings> last_encoding_settings_;
+ std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
+ LimitedTaskQueue task_queue_;
+ std::unique_ptr<TesterY4mWriter> y4m_writer_;
+ std::unique_ptr<TesterIvfWriter> ivf_writer_;
+ std::map<uint32_t, int> sidx_ RTC_GUARDED_BY(mutex_);
+ std::map<uint32_t, EncodeCallback> callbacks_ RTC_GUARDED_BY(mutex_);
+ Mutex mutex_;
+};
+
+std::tuple<std::vector<DataRate>, ScalabilityMode>
+SplitBitrateAndUpdateScalabilityMode(std::string codec_type,
+ ScalabilityMode scalability_mode,
+ int width,
+ int height,
+ std::vector<int> bitrates_kbps,
+ double framerate_fps) {
+ int num_spatial_layers = ScalabilityModeToNumSpatialLayers(scalability_mode);
+ 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";
+ std::vector<DataRate> bitrates;
+ for (const auto& bitrate_kbps : bitrates_kbps) {
+ bitrates.push_back(DataRate::KilobitsPerSec(bitrate_kbps));
+ }
+ return std::make_tuple(bitrates, scalability_mode);
+ }
+
+ VideoCodec vc;
+ vc.codecType = PayloadStringToCodecType(codec_type);
+ vc.width = width;
+ vc.height = height;
+ vc.startBitrate = bitrates_kbps.front();
+ vc.maxBitrate = bitrates_kbps.front();
+ vc.minBitrate = 0;
+ vc.maxFramerate = static_cast<uint32_t>(framerate_fps);
+ vc.numberOfSimulcastStreams = 0;
+ vc.mode = webrtc::VideoCodecMode::kRealtimeVideo;
+ vc.SetScalabilityMode(scalability_mode);
+
+ switch (vc.codecType) {
+ case kVideoCodecVP8:
+ // TODO(webrtc:14852): Configure simulcast.
+ *(vc.VP8()) = VideoEncoder::GetDefaultVp8Settings();
+ vc.VP8()->SetNumberOfTemporalLayers(num_temporal_layers);
+ vc.simulcastStream[0].width = vc.width;
+ vc.simulcastStream[0].height = vc.height;
+ 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);
+ } break;
+ case kVideoCodecH265:
+ 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());
+ }
+
+ std::unique_ptr<VideoBitrateAllocator> bitrate_allocator =
+ CreateBuiltinVideoBitrateAllocatorFactory()->CreateVideoBitrateAllocator(
+ vc);
+ VideoBitrateAllocation bitrate_allocation =
+ bitrate_allocator->Allocate(VideoBitrateAllocationParameters(
+ 1000 * bitrates_kbps.front(), framerate_fps));
+
+ std::vector<DataRate> bitrates;
+ for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+ for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
+ int bitrate_bps = bitrate_allocation.GetBitrate(sidx, tidx);
+ bitrates.push_back(DataRate::BitsPerSec(bitrate_bps));
+ }
+ }
+
+ return std::make_tuple(bitrates, *vc.GetScalabilityMode());
+}
+
+} // namespace
+
+void VideoCodecStats::Stream::LogMetrics(
+ MetricsLogger* logger,
+ std::string test_case_name,
+ std::string prefix,
+ std::map<std::string, std::string> metadata) const {
+ logger->LogMetric(prefix + "width", test_case_name, width, Unit::kCount,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ logger->LogMetric(prefix + "height", test_case_name, height, Unit::kCount,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ logger->LogMetric(prefix + "frame_size_bytes", test_case_name,
+ frame_size_bytes, Unit::kBytes,
+ ImprovementDirection::kNeitherIsBetter, metadata);
+ logger->LogMetric(prefix + "keyframe", test_case_name, keyframe, Unit::kCount,
+ ImprovementDirection::kSmallerIsBetter, metadata);
+ logger->LogMetric(prefix + "qp", test_case_name, qp, Unit::kUnitless,
+ ImprovementDirection::kSmallerIsBetter, metadata);
+ // TODO(webrtc:14852): Change to us or even ns.
+ logger->LogMetric(prefix + "encode_time_ms", test_case_name, encode_time_ms,
+ Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter,
+ metadata);
+ logger->LogMetric(prefix + "decode_time_ms", test_case_name, decode_time_ms,
+ Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter,
+ metadata);
+ // TODO(webrtc:14852): Change to kUnitLess. kKilobitsPerSecond are converted
+ // to bytes per second in Chromeperf dash.
+ logger->LogMetric(prefix + "target_bitrate_kbps", test_case_name,
+ target_bitrate_kbps, Unit::kKilobitsPerSecond,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ logger->LogMetric(prefix + "target_framerate_fps", test_case_name,
+ target_framerate_fps, Unit::kHertz,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ // TODO(webrtc:14852): Change to kUnitLess. kKilobitsPerSecond are converted
+ // to bytes per second in Chromeperf dash.
+ logger->LogMetric(prefix + "encoded_bitrate_kbps", test_case_name,
+ encoded_bitrate_kbps, Unit::kKilobitsPerSecond,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ logger->LogMetric(prefix + "encoded_framerate_fps", test_case_name,
+ encoded_framerate_fps, Unit::kHertz,
+ ImprovementDirection::kBiggerIsBetter, metadata);
+ logger->LogMetric(prefix + "bitrate_mismatch_pct", test_case_name,
+ bitrate_mismatch_pct, Unit::kPercent,
+ ImprovementDirection::kNeitherIsBetter, metadata);
+ logger->LogMetric(prefix + "framerate_mismatch_pct", test_case_name,
+ framerate_mismatch_pct, Unit::kPercent,
+ ImprovementDirection::kNeitherIsBetter, metadata);
+ logger->LogMetric(prefix + "transmission_time_ms", test_case_name,
+ transmission_time_ms, Unit::kMilliseconds,
+ ImprovementDirection::kSmallerIsBetter, metadata);
+ logger->LogMetric(prefix + "psnr_y_db", test_case_name, psnr.y,
+ Unit::kUnitless, ImprovementDirection::kBiggerIsBetter,
+ metadata);
+ logger->LogMetric(prefix + "psnr_u_db", test_case_name, psnr.u,
+ Unit::kUnitless, ImprovementDirection::kBiggerIsBetter,
+ metadata);
+ logger->LogMetric(prefix + "psnr_v_db", test_case_name, psnr.v,
+ Unit::kUnitless, ImprovementDirection::kBiggerIsBetter,
+ metadata);
+}
+
+// TODO(ssilkin): use Frequency and DataRate for framerate and bitrate.
+std::map<uint32_t, EncodingSettings> VideoCodecTester::CreateEncodingSettings(
+ std::string codec_type,
+ std::string scalability_name,
+ int width,
+ int height,
+ std::vector<int> layer_bitrates_kbps,
+ double framerate_fps,
+ int num_frames,
+ uint32_t first_timestamp_rtp) {
+ auto [layer_bitrates, scalability_mode] =
+ SplitBitrateAndUpdateScalabilityMode(
+ codec_type, *ScalabilityModeFromString(scalability_name), width,
+ height, layer_bitrates_kbps, framerate_fps);
+
+ int num_spatial_layers = ScalabilityModeToNumSpatialLayers(scalability_mode);
+ int num_temporal_layers =
+ ScalabilityModeToNumTemporalLayers(scalability_mode);
+
+ std::map<LayerId, LayerSettings> layers_settings;
+ for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+ int layer_width = width >> (num_spatial_layers - sidx - 1);
+ int layer_height = height >> (num_spatial_layers - sidx - 1);
+ for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
+ double layer_framerate_fps =
+ framerate_fps / (1 << (num_temporal_layers - tidx - 1));
+ layers_settings.emplace(
+ LayerId{.spatial_idx = sidx, .temporal_idx = tidx},
+ LayerSettings{
+ .resolution = {.width = layer_width, .height = layer_height},
+ .framerate = Frequency::MilliHertz(1000 * layer_framerate_fps),
+ .bitrate = layer_bitrates[sidx * num_temporal_layers + tidx]});
+ }
+ }
+
+ std::map<uint32_t, EncodingSettings> frames_settings;
+ uint32_t timestamp_rtp = first_timestamp_rtp;
+ for (int frame_num = 0; frame_num < num_frames; ++frame_num) {
+ frames_settings.emplace(
+ timestamp_rtp,
+ EncodingSettings{.sdp_video_format = SdpVideoFormat(codec_type),
+ .scalability_mode = scalability_mode,
+ .layers_settings = layers_settings});
+
+ timestamp_rtp += k90kHz / Frequency::MilliHertz(1000 * framerate_fps);
+ }
+
+ return frames_settings;
+}
+
+std::unique_ptr<VideoCodecTester::VideoCodecStats>
+VideoCodecTester::RunDecodeTest(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.Initialize(sdp_video_format);
+
+ while (auto frame = video_source->PullFrame()) {
+ decoder.Decode(*frame);
+ }
+
+ decoder.Flush();
+ analyzer->Flush();
+ return std::move(analyzer);
+}
+
+std::unique_ptr<VideoCodecTester::VideoCodecStats>
+VideoCodecTester::RunEncodeTest(
+ const VideoSourceSettings& source_settings,
+ VideoEncoderFactory* encoder_factory,
+ const EncoderSettings& encoder_settings,
+ const std::map<uint32_t, EncodingSettings>& encoding_settings) {
+ VideoSource video_source(source_settings);
+ std::unique_ptr<VideoCodecAnalyzer> analyzer =
+ std::make_unique<VideoCodecAnalyzer>(/*video_source=*/nullptr);
+ Encoder encoder(encoder_factory, encoder_settings, analyzer.get());
+ encoder.Initialize(encoding_settings.begin()->second);
+
+ for (const auto& [timestamp_rtp, frame_settings] : encoding_settings) {
+ const EncodingSettings::LayerSettings& top_layer =
+ frame_settings.layers_settings.rbegin()->second;
+ VideoFrame source_frame = video_source.PullFrame(
+ timestamp_rtp, top_layer.resolution, top_layer.framerate);
+ encoder.Encode(source_frame, frame_settings,
+ [](const EncodedImage& encoded_frame) {});
+ }
+
+ encoder.Flush();
+ analyzer->Flush();
+ return std::move(analyzer);
+}
+
+std::unique_ptr<VideoCodecTester::VideoCodecStats>
+VideoCodecTester::RunEncodeDecodeTest(
+ const VideoSourceSettings& source_settings,
+ VideoEncoderFactory* encoder_factory,
+ VideoDecoderFactory* decoder_factory,
+ const EncoderSettings& encoder_settings,
+ const DecoderSettings& decoder_settings,
+ const std::map<uint32_t, EncodingSettings>& encoding_settings) {
+ VideoSource video_source(source_settings);
+ std::unique_ptr<VideoCodecAnalyzer> analyzer =
+ std::make_unique<VideoCodecAnalyzer>(&video_source);
+ Decoder decoder(decoder_factory, decoder_settings, analyzer.get());
+ Encoder encoder(encoder_factory, encoder_settings, analyzer.get());
+ encoder.Initialize(encoding_settings.begin()->second);
+ decoder.Initialize(encoding_settings.begin()->second.sdp_video_format);
+
+ for (const auto& [timestamp_rtp, frame_settings] : encoding_settings) {
+ const EncodingSettings::LayerSettings& top_layer =
+ frame_settings.layers_settings.rbegin()->second;
+ VideoFrame source_frame = video_source.PullFrame(
+ timestamp_rtp, top_layer.resolution, top_layer.framerate);
+ encoder.Encode(source_frame, frame_settings,
+ [&decoder](const EncodedImage& encoded_frame) {
+ decoder.Decode(encoded_frame);
+ });
+ }
+
+ encoder.Flush();
+ decoder.Flush();
+ analyzer->Flush();
+ return std::move(analyzer);
+}
+
+} // namespace test
+} // namespace webrtc
diff --git a/third_party/libwebrtc/test/video_codec_tester.h b/third_party/libwebrtc/test/video_codec_tester.h
new file mode 100644
index 0000000000..87cc5f76f8
--- /dev/null
+++ b/third_party/libwebrtc/test/video_codec_tester.h
@@ -0,0 +1,227 @@
+/*
+ * 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 TEST_VIDEO_CODEC_TESTER_H_
+#define TEST_VIDEO_CODEC_TESTER_H_
+
+#include <limits>
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/frequency.h"
+#include "api/video/encoded_image.h"
+#include "api/video/resolution.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+class VideoCodecTester {
+ public:
+ struct LayerId {
+ int spatial_idx = 0;
+ int temporal_idx = 0;
+
+ bool operator==(const LayerId& o) const {
+ return spatial_idx == o.spatial_idx && temporal_idx == o.temporal_idx;
+ }
+ bool operator<(const LayerId& o) const {
+ return spatial_idx < o.spatial_idx ||
+ (spatial_idx == o.spatial_idx && temporal_idx < o.temporal_idx);
+ }
+ };
+
+ struct EncodingSettings {
+ SdpVideoFormat sdp_video_format = SdpVideoFormat("VP8");
+ ScalabilityMode scalability_mode = ScalabilityMode::kL1T1;
+
+ struct LayerSettings {
+ Resolution resolution;
+ Frequency framerate;
+ DataRate bitrate;
+ };
+ std::map<LayerId, LayerSettings> layers_settings;
+ };
+
+ class VideoCodecStats {
+ public:
+ struct Filter {
+ uint32_t min_timestamp_rtp = std::numeric_limits<uint32_t>::min();
+ uint32_t max_timestamp_rtp = std::numeric_limits<uint32_t>::max();
+ absl::optional<LayerId> layer_id;
+ };
+
+ struct Frame {
+ uint32_t timestamp_rtp = 0;
+ LayerId layer_id;
+ bool encoded = false;
+ bool decoded = false;
+ int width = 0;
+ int height = 0;
+ DataSize frame_size = DataSize::Zero();
+ bool keyframe = false;
+ absl::optional<int> qp;
+ Timestamp encode_start = Timestamp::Zero();
+ TimeDelta encode_time = TimeDelta::Zero();
+ Timestamp decode_start = Timestamp::Zero();
+ TimeDelta decode_time = TimeDelta::Zero();
+ absl::optional<DataRate> target_bitrate;
+ absl::optional<Frequency> target_framerate;
+
+ struct Psnr {
+ double y = 0.0;
+ double u = 0.0;
+ double v = 0.0;
+ };
+ absl::optional<Psnr> psnr;
+ };
+
+ struct Stream {
+ SamplesStatsCounter width;
+ SamplesStatsCounter height;
+ SamplesStatsCounter frame_size_bytes;
+ SamplesStatsCounter keyframe;
+ SamplesStatsCounter qp;
+ SamplesStatsCounter encode_time_ms;
+ SamplesStatsCounter decode_time_ms;
+ SamplesStatsCounter target_bitrate_kbps;
+ SamplesStatsCounter target_framerate_fps;
+ SamplesStatsCounter encoded_bitrate_kbps;
+ SamplesStatsCounter encoded_framerate_fps;
+ SamplesStatsCounter bitrate_mismatch_pct;
+ SamplesStatsCounter framerate_mismatch_pct;
+ SamplesStatsCounter transmission_time_ms;
+
+ struct Psnr {
+ SamplesStatsCounter y;
+ SamplesStatsCounter u;
+ SamplesStatsCounter v;
+ } psnr;
+
+ // Logs `Stream` metrics to provided `MetricsLogger`.
+ void LogMetrics(MetricsLogger* logger,
+ std::string test_case_name,
+ std::string prefix,
+ std::map<std::string, std::string> metadata = {}) const;
+ };
+
+ virtual ~VideoCodecStats() = default;
+
+ // Returns frames for the slice specified by `filter`. If `merge` is true,
+ // also merges frames belonging to the same temporal unit into one
+ // superframe.
+ virtual std::vector<Frame> Slice(Filter filter, bool merge) const = 0;
+
+ // Returns video statistics aggregated for the slice specified by `filter`.
+ virtual Stream Aggregate(Filter filter) const = 0;
+
+ // Write metrics to a CSV file.
+ virtual void LogMetrics(
+ absl::string_view csv_path,
+ std::vector<Frame> frames,
+ std::map<std::string, std::string> metadata) const = 0;
+ };
+
+ // Pacing settings for codec input.
+ struct PacingSettings {
+ enum PacingMode {
+ // Pacing is not used. Frames are sent to codec back-to-back.
+ kNoPacing,
+ // Pace with the rate equal to the target video frame rate. Pacing time is
+ // derived from RTP timestamp.
+ kRealTime,
+ // Pace with the explicitly provided rate.
+ kConstantRate,
+ };
+ PacingMode mode = PacingMode::kNoPacing;
+ // Pacing rate for `kConstantRate` mode.
+ Frequency constant_rate = Frequency::Zero();
+ };
+
+ struct VideoSourceSettings {
+ std::string file_path;
+ Resolution resolution;
+ Frequency framerate;
+ };
+
+ struct DecoderSettings {
+ PacingSettings pacing_settings;
+ absl::optional<std::string> decoder_input_base_path;
+ absl::optional<std::string> decoder_output_base_path;
+ };
+
+ struct EncoderSettings {
+ PacingSettings pacing_settings;
+ absl::optional<std::string> encoder_input_base_path;
+ absl::optional<std::string> encoder_output_base_path;
+ };
+
+ virtual ~VideoCodecTester() = default;
+
+ // Interface for a coded video frames source.
+ class CodedVideoSource {
+ public:
+ virtual ~CodedVideoSource() = default;
+
+ // Returns next frame. Returns `absl::nullopt` if the end-of-stream is
+ // reached. Frames should have RTP timestamps representing desired frame
+ // rate.
+ virtual absl::optional<EncodedImage> PullFrame() = 0;
+ };
+
+ // A helper function that creates `EncodingSettings` for `num_frames` frames,
+ // wraps the settings into RTP timestamp -> settings map and returns the map.
+ static std::map<uint32_t, EncodingSettings> CreateEncodingSettings(
+ std::string codec_type,
+ std::string scalability_name,
+ int width,
+ int height,
+ std::vector<int> bitrates_kbps,
+ double framerate_fps,
+ int num_frames,
+ uint32_t first_timestamp_rtp = 90000);
+
+ // Decodes video, collects and returns decode metrics.
+ static std::unique_ptr<VideoCodecStats> RunDecodeTest(
+ CodedVideoSource* video_source,
+ VideoDecoderFactory* decoder_factory,
+ const DecoderSettings& decoder_settings,
+ const SdpVideoFormat& sdp_video_format);
+
+ // Encodes video, collects and returns encode metrics.
+ static std::unique_ptr<VideoCodecStats> RunEncodeTest(
+ const VideoSourceSettings& source_settings,
+ VideoEncoderFactory* encoder_factory,
+ const EncoderSettings& encoder_settings,
+ const std::map<uint32_t, EncodingSettings>& encoding_settings);
+
+ // Encodes and decodes video, collects and returns encode and decode metrics.
+ static std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
+ const VideoSourceSettings& source_settings,
+ VideoEncoderFactory* encoder_factory,
+ VideoDecoderFactory* decoder_factory,
+ const EncoderSettings& encoder_settings,
+ const DecoderSettings& decoder_settings,
+ const std::map<uint32_t, EncodingSettings>& encoding_settings);
+};
+
+} // namespace test
+} // namespace webrtc
+
+#endif // TEST_VIDEO_CODEC_TESTER_H_
diff --git a/third_party/libwebrtc/test/video_codec_tester_unittest.cc b/third_party/libwebrtc/test/video_codec_tester_unittest.cc
new file mode 100644
index 0000000000..af31fe2c13
--- /dev/null
+++ b/third_party/libwebrtc/test/video_codec_tester_unittest.cc
@@ -0,0 +1,513 @@
+/*
+ * 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 "test/video_codec_tester.h"
+
+#include <map>
+#include <memory>
+#include <string>
+#include <tuple>
+#include <utility>
+#include <vector>
+
+#include "api/test/mock_video_decoder.h"
+#include "api/test/mock_video_decoder_factory.h"
+#include "api/test/mock_video_encoder.h"
+#include "api/test/mock_video_encoder_factory.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "third_party/libyuv/include/libyuv/planar_functions.h"
+
+namespace webrtc {
+namespace test {
+
+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 VideoCodecStats = VideoCodecTester::VideoCodecStats;
+using VideoSourceSettings = VideoCodecTester::VideoSourceSettings;
+using CodedVideoSource = VideoCodecTester::CodedVideoSource;
+using EncodingSettings = VideoCodecTester::EncodingSettings;
+using LayerSettings = EncodingSettings::LayerSettings;
+using LayerId = VideoCodecTester::LayerId;
+using DecoderSettings = VideoCodecTester::DecoderSettings;
+using EncoderSettings = VideoCodecTester::EncoderSettings;
+using PacingSettings = VideoCodecTester::PacingSettings;
+using PacingMode = PacingSettings::PacingMode;
+using Filter = VideoCodecStats::Filter;
+using Frame = VideoCodecTester::VideoCodecStats::Frame;
+using Stream = VideoCodecTester::VideoCodecStats::Stream;
+
+constexpr int kWidth = 2;
+constexpr int kHeight = 2;
+const DataRate kTargetLayerBitrate = DataRate::BytesPerSec(100);
+const Frequency kTargetFramerate = Frequency::Hertz(30);
+constexpr Frequency k90kHz = Frequency::Hertz(90000);
+
+rtc::scoped_refptr<I420Buffer> CreateYuvBuffer(uint8_t y = 0,
+ uint8_t u = 0,
+ uint8_t v = 0) {
+ rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(2, 2));
+
+ libyuv::I420Rect(buffer->MutableDataY(), buffer->StrideY(),
+ buffer->MutableDataU(), buffer->StrideU(),
+ buffer->MutableDataV(), buffer->StrideV(), 0, 0,
+ buffer->width(), buffer->height(), y, u, v);
+ return buffer;
+}
+
+std::string CreateYuvFile(int width, int height, int num_frames) {
+ std::string path = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+ "video_codec_tester_unittest");
+ FILE* file = fopen(path.c_str(), "wb");
+ for (int frame_num = 0; frame_num < num_frames; ++frame_num) {
+ uint8_t y = (frame_num + 0) & 255;
+ uint8_t u = (frame_num + 1) & 255;
+ uint8_t v = (frame_num + 2) & 255;
+ rtc::scoped_refptr<I420Buffer> buffer = CreateYuvBuffer(y, u, v);
+ fwrite(buffer->DataY(), 1, width * height, file);
+ int chroma_size_bytes = (width + 1) / 2 * (height + 1) / 2;
+ fwrite(buffer->DataU(), 1, chroma_size_bytes, file);
+ fwrite(buffer->DataV(), 1, chroma_size_bytes, file);
+ }
+ fclose(file);
+ return path;
+}
+
+std::unique_ptr<VideoCodecStats> RunTest(std::vector<std::vector<Frame>> frames,
+ ScalabilityMode scalability_mode) {
+ int num_frames = static_cast<int>(frames.size());
+ std::string source_yuv_path = CreateYuvFile(kWidth, kHeight, num_frames);
+ VideoSourceSettings source_settings{
+ .file_path = source_yuv_path,
+ .resolution = {.width = kWidth, .height = kHeight},
+ .framerate = kTargetFramerate};
+
+ int num_encoded_frames = 0;
+ EncodedImageCallback* encoded_frame_callback;
+ NiceMock<MockVideoEncoderFactory> encoder_factory;
+ ON_CALL(encoder_factory, CreateVideoEncoder)
+ .WillByDefault([&](const SdpVideoFormat&) {
+ auto encoder = std::make_unique<NiceMock<MockVideoEncoder>>();
+ ON_CALL(*encoder, RegisterEncodeCompleteCallback)
+ .WillByDefault([&](EncodedImageCallback* callback) {
+ encoded_frame_callback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+ });
+ ON_CALL(*encoder, Encode)
+ .WillByDefault([&](const VideoFrame& input_frame,
+ const std::vector<VideoFrameType>*) {
+ for (const Frame& frame : frames[num_encoded_frames]) {
+ EncodedImage encoded_frame;
+ encoded_frame._encodedWidth = frame.width;
+ encoded_frame._encodedHeight = frame.height;
+ encoded_frame.SetFrameType(
+ frame.keyframe ? VideoFrameType::kVideoFrameKey
+ : VideoFrameType::kVideoFrameDelta);
+ encoded_frame.SetRtpTimestamp(input_frame.timestamp());
+ encoded_frame.SetSpatialIndex(frame.layer_id.spatial_idx);
+ encoded_frame.SetTemporalIndex(frame.layer_id.temporal_idx);
+ encoded_frame.SetEncodedData(
+ EncodedImageBuffer::Create(frame.frame_size.bytes()));
+ encoded_frame_callback->OnEncodedImage(
+ encoded_frame,
+ /*codec_specific_info=*/nullptr);
+ }
+ ++num_encoded_frames;
+ return WEBRTC_VIDEO_CODEC_OK;
+ });
+ return encoder;
+ });
+
+ int num_decoded_frames = 0;
+ DecodedImageCallback* decode_callback;
+ NiceMock<MockVideoDecoderFactory> decoder_factory;
+ ON_CALL(decoder_factory, CreateVideoDecoder)
+ .WillByDefault([&](const SdpVideoFormat&) {
+ auto decoder = std::make_unique<NiceMock<MockVideoDecoder>>();
+ ON_CALL(*decoder, RegisterDecodeCompleteCallback)
+ .WillByDefault([&](DecodedImageCallback* callback) {
+ decode_callback = callback;
+ return WEBRTC_VIDEO_CODEC_OK;
+ });
+ ON_CALL(*decoder, Decode(_, _))
+ .WillByDefault([&](const EncodedImage& encoded_frame, int64_t) {
+ // Make values to be different from source YUV generated in
+ // `CreateYuvFile`.
+ uint8_t y = ((num_decoded_frames + 1) * 2) & 255;
+ uint8_t u = ((num_decoded_frames + 2) * 2) & 255;
+ uint8_t v = ((num_decoded_frames + 3) * 2) & 255;
+ rtc::scoped_refptr<I420Buffer> frame_buffer =
+ CreateYuvBuffer(y, u, v);
+ VideoFrame decoded_frame =
+ VideoFrame::Builder()
+ .set_video_frame_buffer(frame_buffer)
+ .set_timestamp_rtp(encoded_frame.RtpTimestamp())
+ .build();
+ decode_callback->Decoded(decoded_frame);
+ ++num_decoded_frames;
+ return WEBRTC_VIDEO_CODEC_OK;
+ });
+ return decoder;
+ });
+
+ int num_spatial_layers = ScalabilityModeToNumSpatialLayers(scalability_mode);
+ int num_temporal_layers =
+ ScalabilityModeToNumTemporalLayers(scalability_mode);
+
+ std::map<uint32_t, EncodingSettings> encoding_settings;
+ for (int frame_num = 0; frame_num < num_frames; ++frame_num) {
+ std::map<LayerId, LayerSettings> layers_settings;
+ for (int sidx = 0; sidx < num_spatial_layers; ++sidx) {
+ for (int tidx = 0; tidx < num_temporal_layers; ++tidx) {
+ layers_settings.emplace(
+ LayerId{.spatial_idx = sidx, .temporal_idx = tidx},
+ LayerSettings{.resolution = {.width = kWidth, .height = kHeight},
+ .framerate = kTargetFramerate /
+ (1 << (num_temporal_layers - 1 - tidx)),
+ .bitrate = kTargetLayerBitrate});
+ }
+ }
+ encoding_settings.emplace(
+ frames[frame_num][0].timestamp_rtp,
+ EncodingSettings{.scalability_mode = scalability_mode,
+ .layers_settings = layers_settings});
+ }
+
+ EncoderSettings encoder_settings;
+ DecoderSettings decoder_settings;
+ std::unique_ptr<VideoCodecStats> stats =
+ VideoCodecTester::RunEncodeDecodeTest(
+ source_settings, &encoder_factory, &decoder_factory, encoder_settings,
+ decoder_settings, encoding_settings);
+ remove(source_yuv_path.c_str());
+ return stats;
+}
+
+EncodedImage CreateEncodedImage(uint32_t timestamp_rtp) {
+ EncodedImage encoded_image;
+ encoded_image.SetRtpTimestamp(timestamp_rtp);
+ return encoded_image;
+}
+
+class MockCodedVideoSource : public CodedVideoSource {
+ public:
+ MockCodedVideoSource(int num_frames, Frequency framerate)
+ : num_frames_(num_frames), frame_num_(0), framerate_(framerate) {}
+
+ absl::optional<EncodedImage> PullFrame() override {
+ if (frame_num_ >= num_frames_) {
+ return absl::nullopt;
+ }
+ uint32_t timestamp_rtp = frame_num_ * k90kHz / framerate_;
+ ++frame_num_;
+ return CreateEncodedImage(timestamp_rtp);
+ }
+
+ private:
+ int num_frames_;
+ int frame_num_;
+ Frequency framerate_;
+};
+
+} // namespace
+
+TEST(VideoCodecTester, Slice) {
+ std::unique_ptr<VideoCodecStats> stats = RunTest(
+ {{{.timestamp_rtp = 0, .layer_id = {.spatial_idx = 0, .temporal_idx = 0}},
+ {.timestamp_rtp = 0,
+ .layer_id = {.spatial_idx = 1, .temporal_idx = 0}}},
+ {{.timestamp_rtp = 1,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 1}}}},
+ ScalabilityMode::kL2T2);
+ std::vector<Frame> slice = stats->Slice(Filter{}, /*merge=*/false);
+ EXPECT_THAT(slice, ElementsAre(Field(&Frame::timestamp_rtp, 0),
+ Field(&Frame::timestamp_rtp, 0),
+ Field(&Frame::timestamp_rtp, 1)));
+
+ slice = stats->Slice({.min_timestamp_rtp = 1}, /*merge=*/false);
+ EXPECT_THAT(slice, ElementsAre(Field(&Frame::timestamp_rtp, 1)));
+
+ slice = stats->Slice({.max_timestamp_rtp = 0}, /*merge=*/false);
+ EXPECT_THAT(slice, ElementsAre(Field(&Frame::timestamp_rtp, 0),
+ Field(&Frame::timestamp_rtp, 0)));
+
+ slice = stats->Slice({.layer_id = {{.spatial_idx = 0, .temporal_idx = 0}}},
+ /*merge=*/false);
+ EXPECT_THAT(slice, ElementsAre(Field(&Frame::timestamp_rtp, 0)));
+
+ slice = stats->Slice({.layer_id = {{.spatial_idx = 0, .temporal_idx = 1}}},
+ /*merge=*/false);
+ EXPECT_THAT(slice, ElementsAre(Field(&Frame::timestamp_rtp, 0),
+ Field(&Frame::timestamp_rtp, 1)));
+}
+
+TEST(VideoCodecTester, Merge) {
+ std::unique_ptr<VideoCodecStats> stats =
+ RunTest({{{.timestamp_rtp = 0,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 0},
+ .frame_size = DataSize::Bytes(1),
+ .keyframe = true},
+ {.timestamp_rtp = 0,
+ .layer_id = {.spatial_idx = 1, .temporal_idx = 0},
+ .frame_size = DataSize::Bytes(2)}},
+ {{.timestamp_rtp = 1,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 1},
+ .frame_size = DataSize::Bytes(4)},
+ {.timestamp_rtp = 1,
+ .layer_id = {.spatial_idx = 1, .temporal_idx = 1},
+ .frame_size = DataSize::Bytes(8)}}},
+ ScalabilityMode::kL2T2_KEY);
+
+ std::vector<Frame> slice = stats->Slice(Filter{}, /*merge=*/true);
+ EXPECT_THAT(
+ slice,
+ ElementsAre(
+ AllOf(Field(&Frame::timestamp_rtp, 0), Field(&Frame::keyframe, true),
+ Field(&Frame::frame_size, DataSize::Bytes(3))),
+ AllOf(Field(&Frame::timestamp_rtp, 1), Field(&Frame::keyframe, false),
+ Field(&Frame::frame_size, DataSize::Bytes(12)))));
+}
+
+struct AggregationTestParameters {
+ Filter filter;
+ double expected_keyframe_sum;
+ double expected_encoded_bitrate_kbps;
+ double expected_encoded_framerate_fps;
+ double expected_bitrate_mismatch_pct;
+ double expected_framerate_mismatch_pct;
+};
+
+class VideoCodecTesterTestAggregation
+ : public ::testing::TestWithParam<AggregationTestParameters> {};
+
+TEST_P(VideoCodecTesterTestAggregation, Aggregate) {
+ AggregationTestParameters test_params = GetParam();
+ std::unique_ptr<VideoCodecStats> stats =
+ RunTest({{// L0T0
+ {.timestamp_rtp = 0,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 0},
+ .frame_size = DataSize::Bytes(1),
+ .keyframe = true},
+ // L1T0
+ {.timestamp_rtp = 0,
+ .layer_id = {.spatial_idx = 1, .temporal_idx = 0},
+ .frame_size = DataSize::Bytes(2)}},
+ // Emulate frame drop (frame_size = 0).
+ {{.timestamp_rtp = 3000,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 0},
+ .frame_size = DataSize::Zero()}},
+ {// L0T1
+ {.timestamp_rtp = 87000,
+ .layer_id = {.spatial_idx = 0, .temporal_idx = 1},
+ .frame_size = DataSize::Bytes(4)},
+ // L1T1
+ {.timestamp_rtp = 87000,
+ .layer_id = {.spatial_idx = 1, .temporal_idx = 1},
+ .frame_size = DataSize::Bytes(8)}}},
+ ScalabilityMode::kL2T2_KEY);
+
+ Stream stream = stats->Aggregate(test_params.filter);
+ EXPECT_EQ(stream.keyframe.GetSum(), test_params.expected_keyframe_sum);
+ EXPECT_EQ(stream.encoded_bitrate_kbps.GetAverage(),
+ test_params.expected_encoded_bitrate_kbps);
+ EXPECT_EQ(stream.encoded_framerate_fps.GetAverage(),
+ test_params.expected_encoded_framerate_fps);
+ EXPECT_EQ(stream.bitrate_mismatch_pct.GetAverage(),
+ test_params.expected_bitrate_mismatch_pct);
+ EXPECT_EQ(stream.framerate_mismatch_pct.GetAverage(),
+ test_params.expected_framerate_mismatch_pct);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ All,
+ VideoCodecTesterTestAggregation,
+ ::testing::Values(
+ // No filtering.
+ AggregationTestParameters{
+ .filter = {},
+ .expected_keyframe_sum = 1,
+ .expected_encoded_bitrate_kbps =
+ DataRate::BytesPerSec(15).kbps<double>(),
+ .expected_encoded_framerate_fps = 2,
+ .expected_bitrate_mismatch_pct =
+ 100 * (15.0 / (kTargetLayerBitrate.bytes_per_sec() * 4) - 1),
+ .expected_framerate_mismatch_pct =
+ 100 * (2.0 / kTargetFramerate.hertz() - 1)},
+ // L0T0
+ AggregationTestParameters{
+ .filter = {.layer_id = {{.spatial_idx = 0, .temporal_idx = 0}}},
+ .expected_keyframe_sum = 1,
+ .expected_encoded_bitrate_kbps =
+ DataRate::BytesPerSec(1).kbps<double>(),
+ .expected_encoded_framerate_fps = 1,
+ .expected_bitrate_mismatch_pct =
+ 100 * (1.0 / kTargetLayerBitrate.bytes_per_sec() - 1),
+ .expected_framerate_mismatch_pct =
+ 100 * (1.0 / (kTargetFramerate.hertz() / 2) - 1)},
+ // L0T1
+ AggregationTestParameters{
+ .filter = {.layer_id = {{.spatial_idx = 0, .temporal_idx = 1}}},
+ .expected_keyframe_sum = 1,
+ .expected_encoded_bitrate_kbps =
+ DataRate::BytesPerSec(5).kbps<double>(),
+ .expected_encoded_framerate_fps = 2,
+ .expected_bitrate_mismatch_pct =
+ 100 * (5.0 / (kTargetLayerBitrate.bytes_per_sec() * 2) - 1),
+ .expected_framerate_mismatch_pct =
+ 100 * (2.0 / kTargetFramerate.hertz() - 1)},
+ // L1T0
+ AggregationTestParameters{
+ .filter = {.layer_id = {{.spatial_idx = 1, .temporal_idx = 0}}},
+ .expected_keyframe_sum = 1,
+ .expected_encoded_bitrate_kbps =
+ DataRate::BytesPerSec(3).kbps<double>(),
+ .expected_encoded_framerate_fps = 1,
+ .expected_bitrate_mismatch_pct =
+ 100 * (3.0 / kTargetLayerBitrate.bytes_per_sec() - 1),
+ .expected_framerate_mismatch_pct =
+ 100 * (1.0 / (kTargetFramerate.hertz() / 2) - 1)},
+ // L1T1
+ AggregationTestParameters{
+ .filter = {.layer_id = {{.spatial_idx = 1, .temporal_idx = 1}}},
+ .expected_keyframe_sum = 1,
+ .expected_encoded_bitrate_kbps =
+ DataRate::BytesPerSec(11).kbps<double>(),
+ .expected_encoded_framerate_fps = 2,
+ .expected_bitrate_mismatch_pct =
+ 100 * (11.0 / (kTargetLayerBitrate.bytes_per_sec() * 2) - 1),
+ .expected_framerate_mismatch_pct =
+ 100 * (2.0 / kTargetFramerate.hertz() - 1)}));
+
+TEST(VideoCodecTester, Psnr) {
+ std::unique_ptr<VideoCodecStats> stats =
+ RunTest({{{.timestamp_rtp = 0, .frame_size = DataSize::Bytes(1)}},
+ {{.timestamp_rtp = 3000, .frame_size = DataSize::Bytes(1)}}},
+ ScalabilityMode::kL1T1);
+
+ std::vector<Frame> slice = stats->Slice(Filter{}, /*merge=*/false);
+ ASSERT_THAT(slice, SizeIs(2));
+ ASSERT_TRUE(slice[0].psnr.has_value());
+ ASSERT_TRUE(slice[1].psnr.has_value());
+ EXPECT_NEAR(slice[0].psnr->y, 42, 1);
+ EXPECT_NEAR(slice[0].psnr->u, 38, 1);
+ EXPECT_NEAR(slice[0].psnr->v, 36, 1);
+ EXPECT_NEAR(slice[1].psnr->y, 38, 1);
+ EXPECT_NEAR(slice[1].psnr->u, 36, 1);
+ EXPECT_NEAR(slice[1].psnr->v, 34, 1);
+}
+
+class VideoCodecTesterTestPacing
+ : public ::testing::TestWithParam<std::tuple<PacingSettings, int>> {
+ public:
+ const int kSourceWidth = 2;
+ const int kSourceHeight = 2;
+ const int kNumFrames = 3;
+ const int kTargetLayerBitrateKbps = 128;
+ const Frequency kTargetFramerate = Frequency::Hertz(10);
+
+ void SetUp() override {
+ source_yuv_file_path_ = webrtc::test::TempFilename(
+ webrtc::test::OutputPath(), "video_codec_tester_impl_unittest");
+ FILE* file = fopen(source_yuv_file_path_.c_str(), "wb");
+ for (int i = 0; i < 3 * kSourceWidth * kSourceHeight / 2; ++i) {
+ fwrite("x", 1, 1, file);
+ }
+ fclose(file);
+ }
+
+ protected:
+ std::string source_yuv_file_path_;
+};
+
+TEST_P(VideoCodecTesterTestPacing, PaceEncode) {
+ auto [pacing_settings, expected_delta_ms] = GetParam();
+ VideoSourceSettings video_source{
+ .file_path = source_yuv_file_path_,
+ .resolution = {.width = kSourceWidth, .height = kSourceHeight},
+ .framerate = kTargetFramerate};
+
+ NiceMock<MockVideoEncoderFactory> encoder_factory;
+ ON_CALL(encoder_factory, CreateVideoEncoder(_))
+ .WillByDefault([](const SdpVideoFormat&) {
+ return std::make_unique<NiceMock<MockVideoEncoder>>();
+ });
+
+ std::map<uint32_t, EncodingSettings> encoding_settings =
+ VideoCodecTester::CreateEncodingSettings(
+ "VP8", "L1T1", kSourceWidth, kSourceHeight, {kTargetLayerBitrateKbps},
+ kTargetFramerate.hertz(), kNumFrames);
+
+ EncoderSettings encoder_settings;
+ encoder_settings.pacing_settings = pacing_settings;
+ std::vector<Frame> frames =
+ VideoCodecTester::RunEncodeTest(video_source, &encoder_factory,
+ encoder_settings, encoding_settings)
+ ->Slice(/*filter=*/{}, /*merge=*/false);
+ ASSERT_THAT(frames, SizeIs(kNumFrames));
+ EXPECT_NEAR((frames[1].encode_start - frames[0].encode_start).ms(),
+ expected_delta_ms, 10);
+ EXPECT_NEAR((frames[2].encode_start - frames[1].encode_start).ms(),
+ expected_delta_ms, 10);
+}
+
+TEST_P(VideoCodecTesterTestPacing, PaceDecode) {
+ auto [pacing_settings, expected_delta_ms] = GetParam();
+ MockCodedVideoSource video_source(kNumFrames, kTargetFramerate);
+
+ NiceMock<MockVideoDecoderFactory> decoder_factory;
+ ON_CALL(decoder_factory, CreateVideoDecoder(_))
+ .WillByDefault([](const SdpVideoFormat&) {
+ 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,
+ decoder_settings, SdpVideoFormat("VP8"))
+ ->Slice(/*filter=*/{}, /*merge=*/false);
+ ASSERT_THAT(frames, SizeIs(kNumFrames));
+ EXPECT_NEAR((frames[1].decode_start - frames[0].decode_start).ms(),
+ expected_delta_ms, 10);
+ EXPECT_NEAR((frames[2].decode_start - frames[1].decode_start).ms(),
+ expected_delta_ms, 10);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ DISABLED_All,
+ VideoCodecTesterTestPacing,
+ ::testing::Values(
+ // No pacing.
+ std::make_tuple(PacingSettings{.mode = PacingMode::kNoPacing},
+ /*expected_delta_ms=*/0),
+ // Real-time pacing.
+ std::make_tuple(PacingSettings{.mode = PacingMode::kRealTime},
+ /*expected_delta_ms=*/100),
+ // Pace with specified constant rate.
+ std::make_tuple(PacingSettings{.mode = PacingMode::kConstantRate,
+ .constant_rate = Frequency::Hertz(20)},
+ /*expected_delta_ms=*/50)));
+} // namespace test
+} // namespace webrtc