Revert "in WebrtcVoiceEngine allow to set TaskQueueFactory"
This reverts commit a39254da593bbdb0b1e072a44827229680afe3ee.
Reason for revert: Tests are failing due to ThreadChecker's called on valid thread.
Original change's description:
> in WebrtcVoiceEngine allow to set TaskQueueFactory
>
> in production code keep using GlobalTaskQueueFactory()
> in tests switch to use DefaultTaskQueueFactory directly.
>
> Bug: webrtc:10284
> Change-Id: I170274a98324796623089a965a39f0cbb7e281d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128878
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27296}
TBR=danilchap@webrtc.org,steveanton@webrtc.org
Change-Id: I9742e5d0171a94f3840e197c40fdb44523e4963b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129780
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27297}diff --git a/media/BUILD.gn b/media/BUILD.gn
index d02ef89..26a17a1 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -260,8 +260,6 @@
libs = []
deps = [
"../api:scoped_refptr",
- "../api/task_queue",
- "../api/task_queue:global_task_queue_factory",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
"../modules/audio_processing:api",
@@ -495,8 +493,6 @@
"../:webrtc_common",
"../api:fake_media_transport",
"../api:scoped_refptr",
- "../api/task_queue",
- "../api/task_queue:default_task_queue_factory",
"../api/test/video:function_video_factory",
"../api/units:time_delta",
"../api/video:video_frame_i420",
diff --git a/media/engine/null_webrtc_video_engine_unittest.cc b/media/engine/null_webrtc_video_engine_unittest.cc
index 343e167..85eaea3 100644
--- a/media/engine/null_webrtc_video_engine_unittest.cc
+++ b/media/engine/null_webrtc_video_engine_unittest.cc
@@ -9,13 +9,7 @@
*/
#include "media/engine/null_webrtc_video_engine.h"
-
-#include <memory>
-#include <utility>
-
#include "absl/memory/memory.h"
-#include "api/task_queue/default_task_queue_factory.h"
-#include "api/task_queue/task_queue_factory.h"
#include "media/engine/webrtc_voice_engine.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
@@ -25,21 +19,30 @@
namespace cricket {
+class WebRtcMediaEngineNullVideo : public CompositeMediaEngine {
+ public:
+ WebRtcMediaEngineNullVideo(
+ webrtc::AudioDeviceModule* adm,
+ const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
+ audio_encoder_factory,
+ const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
+ audio_decoder_factory)
+ : CompositeMediaEngine(absl::make_unique<WebRtcVoiceEngine>(
+ adm,
+ audio_encoder_factory,
+ audio_decoder_factory,
+ nullptr,
+ webrtc::AudioProcessingBuilder().Create()),
+ absl::make_unique<NullWebRtcVideoEngine>()) {}
+};
+
// Simple test to check if NullWebRtcVideoEngine implements the methods
// required by CompositeMediaEngine.
TEST(NullWebRtcVideoEngineTest, CheckInterface) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
- auto audio_engine = absl::make_unique<WebRtcVoiceEngine>(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
- webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
- webrtc::AudioProcessingBuilder().Create());
-
- CompositeMediaEngine engine(std::move(audio_engine),
- absl::make_unique<NullWebRtcVideoEngine>());
-
+ WebRtcMediaEngineNullVideo engine(
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ webrtc::MockAudioDecoderFactory::CreateUnusedFactory());
EXPECT_TRUE(engine.Init());
}
diff --git a/media/engine/webrtc_media_engine.cc b/media/engine/webrtc_media_engine.cc
index 48269ee..5de0927 100644
--- a/media/engine/webrtc_media_engine.cc
+++ b/media/engine/webrtc_media_engine.cc
@@ -14,7 +14,6 @@
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
-#include "api/task_queue/global_task_queue_factory.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
@@ -62,9 +61,9 @@
auto video_engine = absl::make_unique<NullWebRtcVideoEngine>();
#endif
return std::unique_ptr<MediaEngineInterface>(new CompositeMediaEngine(
- absl::make_unique<WebRtcVoiceEngine>(
- &webrtc::GlobalTaskQueueFactory(), adm, audio_encoder_factory,
- audio_decoder_factory, audio_mixer, audio_processing),
+ absl::make_unique<WebRtcVoiceEngine>(adm, audio_encoder_factory,
+ audio_decoder_factory, audio_mixer,
+ audio_processing),
std::move(video_engine)));
}
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index a94fa7a..9fc9d08 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -178,16 +178,12 @@
} // namespace
WebRtcVoiceEngine::WebRtcVoiceEngine(
- webrtc::TaskQueueFactory* task_queue_factory,
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
- : low_priority_worker_queue_(task_queue_factory->CreateTaskQueue(
- "rtc-low-prio",
- webrtc::TaskQueueFactory::Priority::LOW)),
- adm_(adm),
+ : adm_(adm),
encoder_factory_(encoder_factory),
decoder_factory_(decoder_factory),
audio_mixer_(audio_mixer),
@@ -220,6 +216,10 @@
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
+ // TaskQueue expects to be created/destroyed on the same thread.
+ low_priority_worker_queue_.reset(
+ new rtc::TaskQueue("rtc-low-prio", rtc::TaskQueue::Priority::LOW));
+
// Load our audio codec lists.
RTC_LOG(LS_INFO) << "Supported send codecs in order of preference:";
send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
@@ -580,8 +580,8 @@
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- auto aec_dump = webrtc::AecDumpFactory::Create(file, max_size_bytes,
- &low_priority_worker_queue_);
+ auto aec_dump = webrtc::AecDumpFactory::Create(
+ file, max_size_bytes, low_priority_worker_queue_.get());
if (!aec_dump) {
return false;
}
@@ -592,8 +592,8 @@
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
- auto aec_dump =
- webrtc::AecDumpFactory::Create(filename, -1, &low_priority_worker_queue_);
+ auto aec_dump = webrtc::AecDumpFactory::Create(
+ filename, -1, low_priority_worker_queue_.get());
if (aec_dump) {
apm()->AttachAecDump(std::move(aec_dump));
}
diff --git a/media/engine/webrtc_voice_engine.h b/media/engine/webrtc_voice_engine.h
index 1dbba82..3bce78d 100644
--- a/media/engine/webrtc_voice_engine.h
+++ b/media/engine/webrtc_voice_engine.h
@@ -19,7 +19,6 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
-#include "api/task_queue/task_queue_factory.h"
#include "call/audio_state.h"
#include "call/call.h"
#include "media/base/rtp_utils.h"
@@ -47,7 +46,6 @@
public:
WebRtcVoiceEngine(
- webrtc::TaskQueueFactory* task_queue_factory,
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
@@ -97,7 +95,7 @@
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
- rtc::TaskQueue low_priority_worker_queue_;
+ std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_;
webrtc::AudioDeviceModule* adm();
webrtc::AudioProcessing* apm() const;
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 2f6d74a..a8935e1 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -16,7 +16,6 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
-#include "api/task_queue/default_task_queue_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fake_media_engine.h"
@@ -136,8 +135,6 @@
// Tests that our stub library "works".
TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
StrictMock<webrtc::test::MockAudioDeviceModule> adm;
AdmSetupExpectations(&adm);
rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm =
@@ -150,8 +147,7 @@
EXPECT_CALL(*apm, DetachAecDump());
{
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
}
@@ -171,8 +167,7 @@
WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
explicit WebRtcVoiceEngineTestFake(const char* field_trials)
- : task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
- apm_(new rtc::RefCountedObject<
+ : apm_(new rtc::RefCountedObject<
StrictMock<webrtc::test::MockAudioProcessing>>()),
apm_gc_(*apm_->gain_control()),
apm_ns_(*apm_->noise_suppression()),
@@ -203,8 +198,7 @@
auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
engine_.reset(new cricket::WebRtcVoiceEngine(
- task_queue_factory_.get(), &adm_, encoder_factory, decoder_factory,
- nullptr, apm_));
+ &adm_, encoder_factory, decoder_factory, nullptr, apm_));
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
@@ -756,7 +750,6 @@
}
protected:
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
StrictMock<webrtc::test::MockAudioDeviceModule> adm_;
rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_;
webrtc::test::MockGainControl& apm_gc_;
@@ -3499,14 +3492,11 @@
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
// If the VoiceEngine wants to gather available codecs early, that's fine but
// we never want it to create a decoder at this stage.
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
@@ -3521,8 +3511,6 @@
// Tests that reference counting on the external ADM is correct.
TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
EXPECT_CALL(adm, AddRef()).Times(3);
EXPECT_CALL(adm, Release())
@@ -3532,8 +3520,7 @@
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
@@ -3549,16 +3536,13 @@
// Verify the payload id of common audio codecs, including CN, ISAC, and G722.
TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
// TODO(ossu): Why are the payload types of codecs with non-static payload
// type assignments checked here? It shouldn't really matter.
testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
for (const cricket::AudioCodec& codec : engine.send_codecs()) {
@@ -3599,14 +3583,11 @@
// Tests that VoE supports at least 32 channels
TEST(WebRtcVoiceEngineTest, Has32Channels) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
@@ -3634,8 +3615,6 @@
// Test that we set our preferred codecs properly.
TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
// TODO(ossu): I'm not sure of the intent of this test. It's either:
// - Check that our builtin codecs are usable by Channel.
// - The codecs provided by the engine is usable by Channel.
@@ -3647,8 +3626,7 @@
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
cricket::WebRtcVoiceEngine engine(
- task_queue_factory.get(), &adm,
- webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
+ &adm, webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNullImpl event_log;
@@ -3680,8 +3658,6 @@
specs.push_back(
webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}});
- std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
- webrtc::CreateDefaultTaskQueueFactory();
rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory =
webrtc::MockAudioEncoderFactory::CreateUnusedFactory();
rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
@@ -3692,8 +3668,7 @@
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
webrtc::AudioProcessingBuilder().Create();
- cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), &adm,
- unused_encoder_factory,
+ cricket::WebRtcVoiceEngine engine(&adm, unused_encoder_factory,
mock_decoder_factory, nullptr, apm);
engine.Init();
auto codecs = engine.recv_codecs();
