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Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in call tests

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This is practiaclly a reland of the
https://webrtc-review.googlesource.com/c/src/+/157896
except that video multi stream tests are still using the deprecated TaskQueue
(see https://webrtc-review.googlesource.com/c/src/+/159280)

Bug: webrtc:10933
Change-Id: Ie715345924f9dd2d7dd52c99de3ea595b6fad5ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159699
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29794}
This commit is contained in:
Danil Chapovalov 2019-11-13 11:19:53 +01:00 committed by Commit Bot
parent 60bd1aea3d
commit 1b66890a45
3 changed files with 9 additions and 9 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
test/BUILD.gn
View File

@ -748,7 +748,6 @@ rtc_library("test_common") {
":fake_video_codecs",
":fileutils",
":rtp_test_utils",
":single_threaded_task_queue",
":test_support",
":video_test_common",
"../api:rtp_headers",

12
test/call_test.cc
View File

@ -56,7 +56,9 @@ CallTest::CallTest()
num_flexfec_streams_(0),
audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
task_queue_("CallTestTaskQueue") {}
task_queue_(task_queue_factory_->CreateTaskQueue(
"CallTestTaskQueue",
TaskQueueFactory::Priority::NORMAL)) {}
CallTest::~CallTest() = default;
@ -84,7 +86,7 @@ void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
}
void CallTest::RunBaseTest(BaseTest* test) {
SendTask(RTC_FROM_HERE, &task_queue_, [this, test]() {
SendTask(RTC_FROM_HERE, task_queue(), [this, test]() {
num_video_streams_ = test->GetNumVideoStreams();
num_audio_streams_ = test->GetNumAudioStreams();
num_flexfec_streams_ = test->GetNumFlexfecStreams();
@ -123,9 +125,9 @@ void CallTest::RunBaseTest(BaseTest* test) {
CreateReceiverCall(recv_config);
}
test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
receive_transport_ = test->CreateReceiveTransport(&task_queue_);
receive_transport_ = test->CreateReceiveTransport(task_queue());
send_transport_ =
test->CreateSendTransport(&task_queue_, sender_call_.get());
test->CreateSendTransport(task_queue(), sender_call_.get());
if (test->ShouldCreateReceivers()) {
send_transport_->SetReceiver(receiver_call_->Receiver());
@ -184,7 +186,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
test->PerformTest();
SendTask(RTC_FROM_HERE, &task_queue_, [this, test]() {
SendTask(RTC_FROM_HERE, task_queue(), [this, test]() {
Stop();
test->OnStreamsStopped();
DestroyStreams();

5
test/call_test.h
View File

@ -30,7 +30,6 @@
#include "test/fake_vp8_encoder.h"
#include "test/frame_generator_capturer.h"
#include "test/rtp_rtcp_observer.h"
#include "test/single_threaded_task_queue.h"
namespace webrtc {
namespace test {
@ -174,7 +173,7 @@ class CallTest : public ::testing::Test {
void SetVideoEncoderConfig(const VideoEncoderConfig& config);
VideoSendStream* GetVideoSendStream();
FlexfecReceiveStream::Config* GetFlexFecConfig();
TaskQueueBase* task_queue() { return &task_queue_; }
TaskQueueBase* task_queue() { return task_queue_.get(); }
Clock* const clock_;
@ -230,7 +229,7 @@ class CallTest : public ::testing::Test {
void AddRtpExtensionByUri(const std::string& uri,
std::vector<RtpExtension>* extensions) const;
DEPRECATED_SingleThreadedTaskQueueForTesting task_queue_;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
std::vector<RtpExtension> rtp_extensions_;
rtc::scoped_refptr<AudioProcessing> apm_send_;
rtc::scoped_refptr<AudioProcessing> apm_recv_;