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/*
* Copyright (c) 2011 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.
*/
#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
#include <deque>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
class TestErrorObserver;
class LoopBackTransport : public webrtc::Transport {
public:
LoopBackTransport(webrtc::VoENetwork* voe_network, int channel)
: crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()),
packet_event_(webrtc::EventWrapper::Create()),
thread_(webrtc::ThreadWrapper::CreateThread(NetworkProcess,
this,
"LoopBackTransport")),
channel_(channel),
voe_network_(voe_network),
transmitted_packets_(0) {
thread_->Start();
}
~LoopBackTransport() { thread_->Stop(); }
bool SendRtp(const uint8_t* data,
size_t len,
const webrtc::PacketOptions& options) override {
StorePacket(Packet::Rtp, data, len);
return true;
}
bool SendRtcp(const uint8_t* data, size_t len) override {
StorePacket(Packet::Rtcp, data, len);
return true;
}
void WaitForTransmittedPackets(int32_t packet_count) {
enum {
kSleepIntervalMs = 10
};
int32_t limit = transmitted_packets_.Value() + packet_count;
while (transmitted_packets_.Value() < limit) {
webrtc::SleepMs(kSleepIntervalMs);
}
}
void AddChannel(uint32_t ssrc, int channel) {
webrtc::CriticalSectionScoped lock(crit_.get());
channels_[ssrc] = channel;
}
private:
struct Packet {
enum Type { Rtp, Rtcp, } type;
Packet() : len(0) {}
Packet(Type type, const void* data, size_t len)
: type(type), len(len) {
assert(len <= 1500);
memcpy(this->data, data, len);
}
uint8_t data[1500];
size_t len;
};
void StorePacket(Packet::Type type,
const void* data,
size_t len) {
{
webrtc::CriticalSectionScoped lock(crit_.get());
packet_queue_.push_back(Packet(type, data, len));
}
packet_event_->Set();
}
static bool NetworkProcess(void* transport) {
return static_cast<LoopBackTransport*>(transport)->SendPackets();
}
bool SendPackets() {
switch (packet_event_->Wait(10)) {
case webrtc::kEventSignaled:
break;
case webrtc::kEventTimeout:
break;
case webrtc::kEventError:
// TODO(pbos): Log a warning here?
return true;
}
while (true) {
Packet p;
int channel = channel_;
{
webrtc::CriticalSectionScoped lock(crit_.get());
if (packet_queue_.empty())
break;
p = packet_queue_.front();
packet_queue_.pop_front();
if (p.type == Packet::Rtp) {
uint32_t ssrc =
webrtc::ByteReader<uint32_t>::ReadBigEndian(&p.data[8]);
if (channels_[ssrc] != 0)
channel = channels_[ssrc];
}
// TODO(pbos): Add RTCP SSRC muxing/demuxing if anything requires it.
}
// Minimum RTP header size.
if (p.len < 12)
continue;
switch (p.type) {
case Packet::Rtp:
voe_network_->ReceivedRTPPacket(channel, p.data, p.len,
webrtc::PacketTime());
break;
case Packet::Rtcp:
voe_network_->ReceivedRTCPPacket(channel, p.data, p.len);
break;
}
++transmitted_packets_;
}
return true;
}
const rtc::scoped_ptr<webrtc::CriticalSectionWrapper> crit_;
const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
const rtc::scoped_ptr<webrtc::ThreadWrapper> thread_;
std::deque<Packet> packet_queue_ GUARDED_BY(crit_.get());
const int channel_;
std::map<uint32_t, int> channels_ GUARDED_BY(crit_.get());
webrtc::VoENetwork* const voe_network_;
webrtc::Atomic32 transmitted_packets_;
};
// This fixture initializes the voice engine in addition to the work
// done by the before-initialization fixture. It also registers an error
// observer which will fail tests on error callbacks. This fixture is
// useful to tests that want to run before we have started any form of
// streaming through the voice engine.
class AfterInitializationFixture : public BeforeInitializationFixture {
public:
AfterInitializationFixture();
virtual ~AfterInitializationFixture();
protected:
rtc::scoped_ptr<TestErrorObserver> error_observer_;
};
#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
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