| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #include "webrtc/test/direct_transport.h" |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/call.h" |
| #include "webrtc/system_wrappers/interface/clock.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| DirectTransport::DirectTransport() |
| : lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| packet_event_(EventWrapper::Create()), |
| thread_(ThreadWrapper::CreateThread(NetworkProcess, this)), |
| clock_(Clock::GetRealTimeClock()), |
| shutting_down_(false), |
| receiver_(NULL), |
| delay_ms_(0) { |
| unsigned int thread_id; |
| EXPECT_TRUE(thread_->Start(thread_id)); |
| } |
| |
| DirectTransport::DirectTransport(int delay_ms) |
| : lock_(CriticalSectionWrapper::CreateCriticalSection()), |
| packet_event_(EventWrapper::Create()), |
| thread_(ThreadWrapper::CreateThread(NetworkProcess, this)), |
| clock_(Clock::GetRealTimeClock()), |
| shutting_down_(false), |
| receiver_(NULL), |
| delay_ms_(delay_ms) { |
| unsigned int thread_id; |
| EXPECT_TRUE(thread_->Start(thread_id)); |
| } |
| |
| DirectTransport::~DirectTransport() { StopSending(); } |
| |
| void DirectTransport::StopSending() { |
| { |
| CriticalSectionScoped crit_(lock_.get()); |
| shutting_down_ = true; |
| } |
| |
| packet_event_->Set(); |
| EXPECT_TRUE(thread_->Stop()); |
| } |
| |
| void DirectTransport::SetReceiver(PacketReceiver* receiver) { |
| receiver_ = receiver; |
| } |
| |
| bool DirectTransport::SendRTP(const uint8_t* data, size_t length) { |
| QueuePacket(data, length, clock_->TimeInMilliseconds() + delay_ms_); |
| return true; |
| } |
| |
| bool DirectTransport::SendRTCP(const uint8_t* data, size_t length) { |
| QueuePacket(data, length, clock_->TimeInMilliseconds() + delay_ms_); |
| return true; |
| } |
| |
| DirectTransport::Packet::Packet() : length(0), delivery_time_ms(0) {} |
| |
| DirectTransport::Packet::Packet(const uint8_t* data, |
| size_t length, |
| int64_t delivery_time_ms) |
| : length(length), delivery_time_ms(delivery_time_ms) { |
| EXPECT_LE(length, sizeof(this->data)); |
| memcpy(this->data, data, length); |
| } |
| |
| void DirectTransport::QueuePacket(const uint8_t* data, |
| size_t length, |
| int64_t delivery_time_ms) { |
| CriticalSectionScoped crit(lock_.get()); |
| if (receiver_ == NULL) |
| return; |
| packet_queue_.push_back(Packet(data, length, delivery_time_ms)); |
| packet_event_->Set(); |
| } |
| |
| bool DirectTransport::NetworkProcess(void* transport) { |
| return static_cast<DirectTransport*>(transport)->SendPackets(); |
| } |
| |
| bool DirectTransport::SendPackets() { |
| while (true) { |
| Packet p; |
| { |
| CriticalSectionScoped crit(lock_.get()); |
| if (packet_queue_.empty()) |
| break; |
| p = packet_queue_.front(); |
| if (p.delivery_time_ms > clock_->TimeInMilliseconds()) |
| break; |
| packet_queue_.pop_front(); |
| } |
| receiver_->DeliverPacket(p.data, p.length); |
| } |
| uint32_t time_until_next_delivery = WEBRTC_EVENT_INFINITE; |
| { |
| CriticalSectionScoped crit(lock_.get()); |
| if (!packet_queue_.empty()) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| const int64_t delivery_time_ms = packet_queue_.front().delivery_time_ms; |
| if (delivery_time_ms > now_ms) { |
| time_until_next_delivery = delivery_time_ms - now_ms; |
| } else { |
| time_until_next_delivery = 0; |
| } |
| } |
| } |
| |
| switch (packet_event_->Wait(time_until_next_delivery)) { |
| case kEventSignaled: |
| packet_event_->Reset(); |
| break; |
| case kEventTimeout: |
| break; |
| case kEventError: |
| // TODO(pbos): Log a warning here? |
| return true; |
| } |
| |
| CriticalSectionScoped crit(lock_.get()); |
| return shutting_down_ ? false : true; |
| } |
| } // namespace test |
| } // namespace webrtc |