Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced.

Review URL: https://webrtc-codereview.appspot.com/935032

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3259 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-12-11 11:47:22 +00:00
parent f98ffc6db3
commit eaf7cf26fe
4 changed files with 466 additions and 0 deletions

View File

@ -0,0 +1,83 @@
/*
* Copyright (c) 2012 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 WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_
#define WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_
#include <queue>
#include "webrtc/system_wrappers/interface/constructor_magic.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
class NetworkPacket;
class PacketReceiver {
public:
// Delivers a new packet to the receive side of the network pipe. The
// implementor of PacketReceiver now owns the memory.
virtual void IncomingPacket(uint8_t* packet, int length) = 0;
virtual ~PacketReceiver() {}
};
// Class faking a network link. This is a simple and naive solution just faking
// capacity and adding an extra transport delay in addition to the capacity
// introduced delay.
// TODO(mflodman) Add random and bursty packet loss.
class FakeNetworkPipe {
public:
FakeNetworkPipe(PacketReceiver* packet_receiver,
size_t queue_length,
int queue_delay_ms,
int link_capacity_kbps,
int loss_percent);
~FakeNetworkPipe();
// Sends a new packet to the link.
void SendPacket(void* packet, int packet_length);
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void NetworkProcess();
// Get statistics.
float PercentageLoss();
int AverageDelay();
int dropped_packets() { return dropped_packets_; }
int sent_packets() { return sent_packets_; }
private:
PacketReceiver* packet_receiver_;
scoped_ptr<CriticalSectionWrapper> link_cs_;
std::queue<NetworkPacket*> capacity_link_;
std::queue<NetworkPacket*> delay_link_;
// Link configuration.
const size_t queue_length_;
const int queue_delay_ms_;
const int link_capacity_bytes_ms_; // In bytes per ms.
const int loss_percent_;
// Statistics.
int dropped_packets_;
int sent_packets_;
int total_packet_delay_;
DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_

View File

@ -37,12 +37,14 @@
'helpers/vie_to_file_renderer.cc',
# Testbed classes
'include/fake_network_pipe.h',
'include/tb_capture_device.h',
'include/tb_external_transport.h',
'include/tb_I420_codec.h',
'include/tb_interfaces.h',
'include/tb_video_channel.h',
'testbed/fake_network_pipe.cc',
'testbed/tb_capture_device.cc',
'testbed/tb_external_transport.cc',
'testbed/tb_I420_codec.cc',
@ -51,4 +53,26 @@
],
},
],
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'libvietest_unittests',
'type': 'executable',
'dependencies': [
'libvietest',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'include_dirs': [
'include/',
],
'sources': [
'testbed/fake_network_pipe_unittest.cc',
],
},
], #targets
}], # include_tests
], # conditions
}

View File

@ -0,0 +1,154 @@
/*
* Copyright (c) 2012 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/video_engine/test/libvietest/include/fake_network_pipe.h"
#include <assert.h>
#include <string.h>
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
namespace webrtc {
const int kNetworkProcessMaxWaitTime = 10;
class NetworkPacket {
public:
NetworkPacket(void* data, int length, int64_t send_time, int64_t arrival_time)
: data_(NULL),
data_length_(length),
send_time_(send_time),
arrival_time_(arrival_time) {
data_ = new uint8_t[length];
memcpy(data_, data, length);
}
~NetworkPacket() {}
void ReleaseData() {
delete [] data_;
data_ = NULL;
}
uint8_t* data() const { return data_; }
int data_length() const { return data_length_; }
int64_t send_time() const { return send_time_; }
int64_t arrival_time() const { return arrival_time_; }
void IncrementArrivalTime(int64_t extra_delay) {
arrival_time_+= extra_delay;
}
private:
// The packet data.
uint8_t* data_;
// Length of data_.
int data_length_;
// The time the packet was sent out on the network.
const int64_t send_time_;
// The time the packet should arrive at the reciver.
int64_t arrival_time_;
};
FakeNetworkPipe::FakeNetworkPipe(PacketReceiver* packet_receiver,
size_t queue_length, int queue_delay_ms,
int link_capacity_kbps, int loss_percent)
: packet_receiver_(packet_receiver),
link_cs_(CriticalSectionWrapper::CreateCriticalSection()),
queue_length_(queue_length),
queue_delay_ms_(queue_delay_ms),
link_capacity_bytes_ms_(link_capacity_kbps / 8),
loss_percent_(loss_percent),
dropped_packets_(0),
sent_packets_(0),
total_packet_delay_(0) {
assert(link_capacity_bytes_ms_ > 0);
}
FakeNetworkPipe::~FakeNetworkPipe() {
}
void FakeNetworkPipe::SendPacket(void* data, int data_length) {
CriticalSectionScoped cs(link_cs_.get());
if (capacity_link_.size() >= queue_length_) {
// Too many packet on the link, drop this one.
++dropped_packets_;
return;
}
int64_t time_now = TickTime::MillisecondTimestamp();
// Delay introduced by the link capacity.
int64_t capacity_delay_ms = data_length / link_capacity_bytes_ms_;
int64_t network_start_time = time_now;
// Check if there already are packets on the link and change network start
// time if there is.
if (capacity_link_.size() > 0)
network_start_time = capacity_link_.back()->arrival_time();
int64_t arrival_time = network_start_time + capacity_delay_ms;
NetworkPacket* packet = new NetworkPacket(data, data_length, time_now,
arrival_time);
capacity_link_.push(packet);
}
float FakeNetworkPipe::PercentageLoss() {
CriticalSectionScoped cs(link_cs_.get());
if (sent_packets_ == 0)
return 0;
return static_cast<float>(dropped_packets_) /
(sent_packets_ + dropped_packets_);
}
int FakeNetworkPipe::AverageDelay() {
CriticalSectionScoped cs(link_cs_.get());
if (sent_packets_ == 0)
return 0;
return total_packet_delay_ / sent_packets_;
}
void FakeNetworkPipe::NetworkProcess() {
CriticalSectionScoped cs(link_cs_.get());
if (capacity_link_.size() == 0 && delay_link_.size() == 0)
return;
int64_t time_now = TickTime::MillisecondTimestamp();
// Check the capacity link first.
while (capacity_link_.size() > 0 &&
time_now >= capacity_link_.front()->arrival_time()) {
// Time to get this packet.
NetworkPacket* packet = capacity_link_.front();
capacity_link_.pop();
// Add the packet to the extra delay queue.
packet->IncrementArrivalTime(queue_delay_ms_);
delay_link_.push(packet);
}
// Check the extra delay queue.
while (delay_link_.size() > 0 &&
time_now >= delay_link_.front()->arrival_time()) {
// Deliver this packet.
NetworkPacket* packet = delay_link_.front();
delay_link_.pop();
packet_receiver_->IncomingPacket(packet->data(), packet->data_length());
++sent_packets_;
// |time_now| might be later than when the packet should have arrived, due
// to NetworkProcess being called too late. For stats, use the time it
// should have been on the link.
total_packet_delay_ += packet->arrival_time() - packet->send_time();
delete packet;
}
}
} // namespace webrtc

View File

@ -0,0 +1,205 @@
/*
* Copyright (c) 2012 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 <gmock/gmock.h>
#include <gtest/gtest.h>
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/video_engine/test/libvietest/include/fake_network_pipe.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::Return;
using ::testing::Invoke;
namespace webrtc {
class MockReceiver : public PacketReceiver {
public:
MockReceiver() {}
virtual ~MockReceiver() {}
void IncomingPacket(uint8_t* data, int length) {
IncomingData(data, length);
delete [] data;
}
MOCK_METHOD2(IncomingData, void(uint8_t*, int));
};
class FakeNetworkPipeTest : public ::testing::Test {
protected:
virtual void SetUp() {
TickTime::UseFakeClock(12345);
receiver_.reset(new MockReceiver());
}
virtual void TearDown() {
}
void SendPackets(FakeNetworkPipe* pipe, int number_packets, int kPacketSize) {
scoped_array<uint8_t> packet(new uint8_t[kPacketSize]);
for (int i = 0; i < number_packets; ++i) {
pipe->SendPacket(packet.get(), kPacketSize);
}
}
int PacketTimeMs(int capacity_kbps, int kPacketSize) {
return 8 * kPacketSize / capacity_kbps;
}
scoped_ptr<MockReceiver> receiver_;
};
void DeleteMemory(uint8_t* data, int length) { delete [] data; }
// Test the capacity link and verify we get as many packets as we expect.
TEST_F(FakeNetworkPipeTest, CapacityTest) {
const int kQueueLength = 20;
const int kNetworkDelayMs = 0;
const int kLinkCapacityKbps = 80;
const int kLossPercent = 0;
scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
kQueueLength,
kNetworkDelayMs,
kLinkCapacityKbps,
kLossPercent));
// Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
// get through the pipe.
const int kNumPackets = 10;
const int kPacketSize = 1000;
SendPackets(pipe.get(), kNumPackets , kPacketSize);
// Time to get one packet through the link.
const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
// Time haven't increased yet, so we souldn't get any packets.
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(0);
pipe->NetworkProcess();
// Advance enough time to release one packet.
TickTime::AdvanceFakeClock(kPacketTimeMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(1);
pipe->NetworkProcess();
// Release all but one packet
TickTime::AdvanceFakeClock(9 * kPacketTimeMs - 1);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(8);
pipe->NetworkProcess();
// And the last one.
TickTime::AdvanceFakeClock(1);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(1);
pipe->NetworkProcess();
}
// Test the extra network delay.
TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
const int kQueueLength = 20;
const int kNetworkDelayMs = 100;
const int kLinkCapacityKbps = 80;
const int kLossPercent = 0;
scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
kQueueLength,
kNetworkDelayMs,
kLinkCapacityKbps,
kLossPercent));
const int kNumPackets = 2;
const int kPacketSize = 1000;
SendPackets(pipe.get(), kNumPackets , kPacketSize);
// Time to get one packet through the link.
const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
// Increase more than kPacketTimeMs, but not more than the extra delay.
TickTime::AdvanceFakeClock(kPacketTimeMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(0);
pipe->NetworkProcess();
// Advance the network delay to get the first packet.
TickTime::AdvanceFakeClock(kNetworkDelayMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(1);
pipe->NetworkProcess();
// Advance one more kPacketTimeMs to get the last packet.
TickTime::AdvanceFakeClock(kPacketTimeMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(1);
pipe->NetworkProcess();
}
// Test the number of buffers and packets are dropped when sending too many
// packets too quickly.
TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
const int kQueueLength = 2;
const int kNetworkDelayMs = 0;
const int kLinkCapacityKbps = 80;
const int kLossPercent = 0;
scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
kQueueLength,
kNetworkDelayMs,
kLinkCapacityKbps,
kLossPercent));
const int kPacketSize = 1000;
const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
// Send three packets and verify only 2 are delivered.
SendPackets(pipe.get(), 3, kPacketSize);
// Increase time enough to deliver all three packets, verify only two are
// delivered.
TickTime::AdvanceFakeClock(3 * kPacketTimeMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(2);
pipe->NetworkProcess();
}
// Test we get statistics as expected.
TEST_F(FakeNetworkPipeTest, StatisticsTest) {
const int kQueueLength = 2;
const int kNetworkDelayMs = 20;
const int kLinkCapacityKbps = 80;
const int kLossPercent = 0;
scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
kQueueLength,
kNetworkDelayMs,
kLinkCapacityKbps,
kLossPercent));
const int kPacketSize = 1000;
const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
// Send three packets and verify only 2 are delivered.
SendPackets(pipe.get(), 3, kPacketSize);
TickTime::AdvanceFakeClock(3 * kPacketTimeMs + kNetworkDelayMs);
EXPECT_CALL(*receiver_, IncomingData(_, _))
.Times(2);
pipe->NetworkProcess();
// Packet 1: kPacketTimeMs + kNetworkDelayMs, packet 2: 2 * kPacketTimeMs +
// kNetworkDelayMs => 170 ms average.
EXPECT_EQ(pipe->AverageDelay(), 170);
EXPECT_EQ(pipe->sent_packets(), 2);
EXPECT_EQ(pipe->dropped_packets(), 1);
EXPECT_EQ(pipe->PercentageLoss(), 1/3.f);
}
} // namespace webrtc