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Delete AsyncInvoker usage from SimulatedPacketTransport

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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}
This commit is contained in:
Niels Möller
2021-04-15 11:42:50 +02:00
committed by Commit Bot
parent bc959b61b3
commit affd2196a9
2 changed files with 13 additions and 21 deletions
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1
media/BUILD.gn
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@ -647,6 +647,7 @@ if (rtc_include_tests) {
"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",
]

33
media/sctp/usrsctp_transport_reliability_unittest.cc
View File

@ -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 @@ class SimulatedPacketTransport final : public rtc::PacketTransportInternal {
~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 @@ class SimulatedPacketTransport final : public rtc::PacketTransportInternal {
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 @@ class SimulatedPacketTransport final : public rtc::PacketTransportInternal {
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 @@ class SimulatedPacketTransport final : public rtc::PacketTransportInternal {
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);
};