Make TaskQueueFactory required construction parameter for Call
Bug: webrtc:10284 Change-Id: I573ee0087c035e26918260c21b8b0213ddfe7ebc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143791 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28467}
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@ -227,7 +227,6 @@ rtc_static_library("call") {
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"../api:rtp_headers",
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"../api:simulated_network_api",
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"../api:transport_api",
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"../api/task_queue:global_task_queue_factory",
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"../api/transport:network_control",
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"../api/units:time_delta",
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"../api/video_codecs:video_codecs_api",
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11
call/call.cc
11
call/call.cc
@ -18,7 +18,6 @@
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#include "absl/memory/memory.h"
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#include "absl/types/optional.h"
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#include "api/task_queue/global_task_queue_factory.h"
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#include "api/transport/network_control.h"
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#include "audio/audio_receive_stream.h"
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#include "audio/audio_send_stream.h"
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@ -430,18 +429,14 @@ Call* Call::Create(const Call::Config& config,
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Clock* clock,
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std::unique_ptr<ProcessThread> call_thread,
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std::unique_ptr<ProcessThread> pacer_thread) {
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// TODO(bugs.webrtc.org/10284): DCHECK task_queue_factory dependency is
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// always provided in the config.
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TaskQueueFactory* task_queue_factory = config.task_queue_factory
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? config.task_queue_factory
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: &GlobalTaskQueueFactory();
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RTC_DCHECK(config.task_queue_factory);
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return new internal::Call(
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clock, config,
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absl::make_unique<RtpTransportControllerSend>(
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clock, config.event_log, config.network_state_predictor_factory,
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config.network_controller_factory, config.bitrate_config,
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std::move(pacer_thread), task_queue_factory),
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std::move(call_thread), task_queue_factory);
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std::move(pacer_thread), config.task_queue_factory),
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std::move(call_thread), config.task_queue_factory);
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}
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// This method here to avoid subclasses has to implement this method.
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@ -47,7 +47,7 @@ struct CallConfig {
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// FecController to use for this call.
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FecControllerFactoryInterface* fec_controller_factory = nullptr;
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// Task Queue Factory to be used in this call.
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// Task Queue Factory to be used in this call. Required.
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TaskQueueFactory* task_queue_factory = nullptr;
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// NetworkStatePredictor to use for this call.
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@ -15,6 +15,7 @@
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#include "absl/memory/memory.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/test/fake_media_transport.h"
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#include "api/test/mock_audio_mixer.h"
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#include "audio/audio_receive_stream.h"
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@ -35,6 +36,7 @@ namespace {
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struct CallHelper {
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CallHelper() {
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task_queue_factory_ = webrtc::CreateDefaultTaskQueueFactory();
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webrtc::AudioState::Config audio_state_config;
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audio_state_config.audio_mixer =
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new rtc::RefCountedObject<webrtc::test::MockAudioMixer>();
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@ -44,6 +46,7 @@ struct CallHelper {
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new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>();
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webrtc::Call::Config config(&event_log_);
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config.audio_state = webrtc::AudioState::Create(audio_state_config);
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config.task_queue_factory = task_queue_factory_.get();
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call_.reset(webrtc::Call::Create(config));
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}
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@ -51,6 +54,7 @@ struct CallHelper {
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private:
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webrtc::RtcEventLogNullImpl event_log_;
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std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
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std::unique_ptr<webrtc::Call> call_;
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};
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} // namespace
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@ -19,6 +19,7 @@
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#include "absl/strings/match.h"
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#include "api/media_transport_config.h"
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#include "api/rtp_parameters.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/test/fake_media_transport.h"
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#include "api/test/mock_video_bitrate_allocator.h"
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#include "api/test/mock_video_bitrate_allocator_factory.h"
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@ -226,7 +227,12 @@ class WebRtcVideoEngineTest : public ::testing::Test {
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? nullptr
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: absl::make_unique<webrtc::test::ScopedFieldTrials>(
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field_trials)),
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call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
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task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
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call_(webrtc::Call::Create([&] {
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webrtc::Call::Config call_config(&event_log_);
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call_config.task_queue_factory = task_queue_factory_.get();
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return call_config;
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}())),
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encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
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decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
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video_bitrate_allocator_factory_(
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@ -264,6 +270,7 @@ class WebRtcVideoEngineTest : public ::testing::Test {
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rtc::ScopedFakeClock fake_clock_;
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std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_;
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webrtc::RtcEventLogNullImpl event_log_;
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std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
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// Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
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// initialized when the constructor is called.
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std::unique_ptr<webrtc::Call> call_;
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@ -1137,8 +1144,10 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
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// Create a call.
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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// Create send channel.
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const int send_ssrc = 123;
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@ -1205,8 +1214,10 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
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// Create a call.
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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const auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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// Create recv channel.
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const int recv_ssrc = 321;
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@ -1286,7 +1297,8 @@ TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
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class WebRtcVideoChannelBaseTest : public ::testing::Test {
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protected:
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WebRtcVideoChannelBaseTest()
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: video_bitrate_allocator_factory_(
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: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
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video_bitrate_allocator_factory_(
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webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
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engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
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webrtc::CreateBuiltinVideoDecoderFactory()) {}
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@ -1294,7 +1306,9 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
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virtual void SetUp() {
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// One testcase calls SetUp in a loop, only create call_ once.
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if (!call_) {
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call_.reset(webrtc::Call::Create(webrtc::Call::Config(&event_log_)));
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webrtc::Call::Config call_config(&event_log_);
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call_config.task_queue_factory = task_queue_factory_.get();
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call_.reset(webrtc::Call::Create(call_config));
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}
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cricket::MediaConfig media_config;
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// Disabling cpu overuse detection actually disables quality scaling too; it
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@ -1475,6 +1489,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
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}
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webrtc::RtcEventLogNullImpl event_log_;
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std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
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std::unique_ptr<webrtc::Call> call_;
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std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
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video_bitrate_allocator_factory_;
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@ -11,6 +11,7 @@
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#include <memory>
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#include <utility>
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#include "absl/memory/memory.h"
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#include "absl/strings/match.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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@ -3457,8 +3458,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
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webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
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engine.Init();
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
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call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
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webrtc::CryptoOptions());
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@ -3484,8 +3486,9 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
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webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
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engine.Init();
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
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call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
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webrtc::CryptoOptions());
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@ -3557,8 +3560,9 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
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webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
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engine.Init();
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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cricket::VoiceMediaChannel* channels[32];
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size_t num_channels = 0;
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@ -3599,8 +3603,9 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
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webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
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engine.Init();
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<webrtc::Call> call(
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webrtc::Call::Create(webrtc::Call::Config(&event_log)));
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webrtc::Call::Config call_config(&event_log);
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call_config.task_queue_factory = task_queue_factory.get();
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auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
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cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
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cricket::AudioOptions(),
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webrtc::CryptoOptions(), call.get());
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@ -22,6 +22,7 @@
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#include "api/peer_connection_interface.h"
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#include "api/peer_connection_proxy.h"
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#include "api/scoped_refptr.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/test/fake_media_transport.h"
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#include "media/base/codec.h"
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#include "media/base/fake_media_engine.h"
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@ -71,6 +72,7 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies(
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deps.network_thread = network_thread;
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deps.worker_thread = worker_thread;
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deps.signaling_thread = signaling_thread;
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deps.task_queue_factory = CreateDefaultTaskQueueFactory();
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deps.media_engine = std::move(media_engine);
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deps.call_factory = std::move(call_factory);
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deps.media_transport_factory = std::move(media_transport_factory);
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@ -22,6 +22,7 @@
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#include "api/peer_connection_proxy.h"
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#include "api/rtc_error.h"
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#include "api/scoped_refptr.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "media/base/fake_media_engine.h"
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#include "p2p/base/mock_async_resolver.h"
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#include "p2p/base/port_allocator.h"
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@ -74,6 +75,7 @@ class PeerConnectionFactoryForUsageHistogramTest
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dependencies.network_thread = rtc::Thread::Current();
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dependencies.worker_thread = rtc::Thread::Current();
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dependencies.signaling_thread = rtc::Thread::Current();
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dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
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dependencies.media_engine =
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absl::make_unique<cricket::FakeMediaEngine>();
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dependencies.call_factory = CreateCallFactory();
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@ -217,6 +217,7 @@ void CallTest::CreateSenderCall() {
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void CallTest::CreateSenderCall(const Call::Config& config) {
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auto sender_config = config;
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sender_config.task_queue_factory = task_queue_factory_.get();
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sender_config.network_state_predictor_factory =
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network_state_predictor_factory_.get();
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sender_config.network_controller_factory = network_controller_factory_.get();
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@ -224,7 +225,9 @@ void CallTest::CreateSenderCall(const Call::Config& config) {
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}
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void CallTest::CreateReceiverCall(const Call::Config& config) {
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receiver_call_.reset(Call::Create(config));
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auto receiver_config = config;
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receiver_config.task_queue_factory = task_queue_factory_.get();
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receiver_call_.reset(Call::Create(receiver_config));
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}
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void CallTest::DestroyCalls() {
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@ -48,6 +48,7 @@ void MultiStreamTester::RunTest() {
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webrtc::RtcEventLogNullImpl event_log;
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auto task_queue_factory = CreateDefaultTaskQueueFactory();
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Call::Config config(&event_log);
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config.task_queue_factory = task_queue_factory.get();
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std::unique_ptr<Call> sender_call;
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std::unique_ptr<Call> receiver_call;
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std::unique_ptr<test::DirectTransport> sender_transport;
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