Delete AsyncInvoker usage from SimulatedPacketTransport
Bug: webrtc:12339
Change-Id: Ic293f9c8791ec24025f9eac39cbc4fcf2583d3ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212867
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33741}
diff --git a/media/BUILD.gn b/media/BUILD.gn
index eea3c9a..af59b59 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -647,6 +647,7 @@
"sctp/usrsctp_transport_unittest.cc",
]
deps += [
+ "../rtc_base:rtc_event",
"../rtc_base/task_utils:pending_task_safety_flag",
"../rtc_base/task_utils:to_queued_task",
]
diff --git a/media/sctp/usrsctp_transport_reliability_unittest.cc b/media/sctp/usrsctp_transport_reliability_unittest.cc
index 98f04a4..ddc8419 100644
--- a/media/sctp/usrsctp_transport_reliability_unittest.cc
+++ b/media/sctp/usrsctp_transport_reliability_unittest.cc
@@ -13,8 +13,8 @@
#include "media/sctp/sctp_transport_internal.h"
#include "media/sctp/usrsctp_transport.h"
-#include "rtc_base/async_invoker.h"
#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/event.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/random.h"
@@ -54,11 +54,6 @@
~SimulatedPacketTransport() override {
RTC_DCHECK_RUN_ON(transport_thread_);
- auto destination = destination_.load();
- if (destination != nullptr) {
- invoker_.Flush(destination->transport_thread_);
- }
- invoker_.Flush(transport_thread_);
destination_ = nullptr;
SignalWritableState(this);
}
@@ -83,15 +78,13 @@
return 0;
}
rtc::CopyOnWriteBuffer buffer(data, len);
- auto send_job = [this, flags, buffer = std::move(buffer)] {
- auto destination = destination_.load();
- if (destination == nullptr) {
- return;
- }
- destination->SignalReadPacket(
- destination, reinterpret_cast<const char*>(buffer.data()),
- buffer.size(), rtc::Time(), flags);
- };
+ auto send_task = ToQueuedTask(
+ destination->task_safety_.flag(),
+ [destination, flags, buffer = std::move(buffer)] {
+ destination->SignalReadPacket(
+ destination, reinterpret_cast<const char*>(buffer.data()),
+ buffer.size(), rtc::Time(), flags);
+ });
// Introduce random send delay in range [0 .. 2 * avg_send_delay_millis_]
// millis, which will also work as random packet reordering mechanism.
uint16_t actual_send_delay = avg_send_delay_millis_;
@@ -101,12 +94,10 @@
actual_send_delay += reorder_delay;
if (actual_send_delay > 0) {
- invoker_.AsyncInvokeDelayed<void>(RTC_FROM_HERE,
- destination->transport_thread_,
- std::move(send_job), actual_send_delay);
+ destination->transport_thread_->PostDelayedTask(std::move(send_task),
+ actual_send_delay);
} else {
- invoker_.AsyncInvoke<void>(RTC_FROM_HERE, destination->transport_thread_,
- std::move(send_job));
+ destination->transport_thread_->PostTask(std::move(send_task));
}
return 0;
}
@@ -136,8 +127,8 @@
const uint8_t packet_loss_percents_;
const uint16_t avg_send_delay_millis_;
std::atomic<SimulatedPacketTransport*> destination_ ATOMIC_VAR_INIT(nullptr);
- rtc::AsyncInvoker invoker_;
webrtc::Random random_;
+ webrtc::ScopedTaskSafety task_safety_;
RTC_DISALLOW_COPY_AND_ASSIGN(SimulatedPacketTransport);
};