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
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@ -0,0 +1,83 @@
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_
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#define WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_
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#include <queue>
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#include "webrtc/system_wrappers/interface/constructor_magic.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class NetworkPacket;
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class PacketReceiver {
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public:
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// Delivers a new packet to the receive side of the network pipe. The
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// implementor of PacketReceiver now owns the memory.
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virtual void IncomingPacket(uint8_t* packet, int length) = 0;
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virtual ~PacketReceiver() {}
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};
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// Class faking a network link. This is a simple and naive solution just faking
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// capacity and adding an extra transport delay in addition to the capacity
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// introduced delay.
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// TODO(mflodman) Add random and bursty packet loss.
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class FakeNetworkPipe {
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public:
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FakeNetworkPipe(PacketReceiver* packet_receiver,
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size_t queue_length,
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int queue_delay_ms,
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int link_capacity_kbps,
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int loss_percent);
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~FakeNetworkPipe();
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// Sends a new packet to the link.
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void SendPacket(void* packet, int packet_length);
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// Processes the network queues and trigger PacketReceiver::IncomingPacket for
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// packets ready to be delivered.
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void NetworkProcess();
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// Get statistics.
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float PercentageLoss();
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int AverageDelay();
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int dropped_packets() { return dropped_packets_; }
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int sent_packets() { return sent_packets_; }
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private:
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PacketReceiver* packet_receiver_;
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scoped_ptr<CriticalSectionWrapper> link_cs_;
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std::queue<NetworkPacket*> capacity_link_;
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std::queue<NetworkPacket*> delay_link_;
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// Link configuration.
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const size_t queue_length_;
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const int queue_delay_ms_;
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const int link_capacity_bytes_ms_; // In bytes per ms.
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const int loss_percent_;
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// Statistics.
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int dropped_packets_;
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int sent_packets_;
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int total_packet_delay_;
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DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_TEST_LIBVIETEST_INCLUDE_FAKE_NETWORK_PIPE_H_
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@ -37,12 +37,14 @@
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'helpers/vie_to_file_renderer.cc',
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# Testbed classes
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'include/fake_network_pipe.h',
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'include/tb_capture_device.h',
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'include/tb_external_transport.h',
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'include/tb_I420_codec.h',
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'include/tb_interfaces.h',
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'include/tb_video_channel.h',
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'testbed/fake_network_pipe.cc',
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'testbed/tb_capture_device.cc',
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'testbed/tb_external_transport.cc',
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'testbed/tb_I420_codec.cc',
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@ -51,4 +53,26 @@
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],
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},
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],
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'conditions': [
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['include_tests==1', {
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'targets': [
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{
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'target_name': 'libvietest_unittests',
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'type': 'executable',
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'dependencies': [
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'libvietest',
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'<(DEPTH)/testing/gtest.gyp:gtest',
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'<(DEPTH)/testing/gmock.gyp:gmock',
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'<(webrtc_root)/test/test.gyp:test_support_main',
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],
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'include_dirs': [
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'include/',
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],
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'sources': [
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'testbed/fake_network_pipe_unittest.cc',
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],
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},
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], #targets
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}], # include_tests
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], # conditions
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}
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154
webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
Normal file
154
webrtc/video_engine/test/libvietest/testbed/fake_network_pipe.cc
Normal file
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/test/libvietest/include/fake_network_pipe.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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namespace webrtc {
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const int kNetworkProcessMaxWaitTime = 10;
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class NetworkPacket {
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public:
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NetworkPacket(void* data, int length, int64_t send_time, int64_t arrival_time)
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: data_(NULL),
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data_length_(length),
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send_time_(send_time),
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arrival_time_(arrival_time) {
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data_ = new uint8_t[length];
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memcpy(data_, data, length);
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}
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~NetworkPacket() {}
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void ReleaseData() {
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delete [] data_;
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data_ = NULL;
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}
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uint8_t* data() const { return data_; }
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int data_length() const { return data_length_; }
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int64_t send_time() const { return send_time_; }
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int64_t arrival_time() const { return arrival_time_; }
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void IncrementArrivalTime(int64_t extra_delay) {
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arrival_time_+= extra_delay;
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}
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private:
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// The packet data.
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uint8_t* data_;
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// Length of data_.
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int data_length_;
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// The time the packet was sent out on the network.
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const int64_t send_time_;
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// The time the packet should arrive at the reciver.
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int64_t arrival_time_;
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};
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FakeNetworkPipe::FakeNetworkPipe(PacketReceiver* packet_receiver,
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size_t queue_length, int queue_delay_ms,
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int link_capacity_kbps, int loss_percent)
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: packet_receiver_(packet_receiver),
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link_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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queue_length_(queue_length),
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queue_delay_ms_(queue_delay_ms),
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link_capacity_bytes_ms_(link_capacity_kbps / 8),
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loss_percent_(loss_percent),
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dropped_packets_(0),
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sent_packets_(0),
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total_packet_delay_(0) {
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assert(link_capacity_bytes_ms_ > 0);
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}
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FakeNetworkPipe::~FakeNetworkPipe() {
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}
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void FakeNetworkPipe::SendPacket(void* data, int data_length) {
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CriticalSectionScoped cs(link_cs_.get());
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if (capacity_link_.size() >= queue_length_) {
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// Too many packet on the link, drop this one.
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++dropped_packets_;
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return;
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}
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int64_t time_now = TickTime::MillisecondTimestamp();
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// Delay introduced by the link capacity.
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int64_t capacity_delay_ms = data_length / link_capacity_bytes_ms_;
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int64_t network_start_time = time_now;
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// Check if there already are packets on the link and change network start
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// time if there is.
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if (capacity_link_.size() > 0)
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network_start_time = capacity_link_.back()->arrival_time();
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int64_t arrival_time = network_start_time + capacity_delay_ms;
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NetworkPacket* packet = new NetworkPacket(data, data_length, time_now,
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arrival_time);
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capacity_link_.push(packet);
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}
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float FakeNetworkPipe::PercentageLoss() {
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CriticalSectionScoped cs(link_cs_.get());
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if (sent_packets_ == 0)
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return 0;
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return static_cast<float>(dropped_packets_) /
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(sent_packets_ + dropped_packets_);
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}
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int FakeNetworkPipe::AverageDelay() {
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CriticalSectionScoped cs(link_cs_.get());
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if (sent_packets_ == 0)
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return 0;
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return total_packet_delay_ / sent_packets_;
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}
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void FakeNetworkPipe::NetworkProcess() {
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CriticalSectionScoped cs(link_cs_.get());
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if (capacity_link_.size() == 0 && delay_link_.size() == 0)
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return;
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int64_t time_now = TickTime::MillisecondTimestamp();
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// Check the capacity link first.
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while (capacity_link_.size() > 0 &&
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time_now >= capacity_link_.front()->arrival_time()) {
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// Time to get this packet.
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NetworkPacket* packet = capacity_link_.front();
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capacity_link_.pop();
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// Add the packet to the extra delay queue.
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packet->IncrementArrivalTime(queue_delay_ms_);
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delay_link_.push(packet);
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}
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// Check the extra delay queue.
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while (delay_link_.size() > 0 &&
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time_now >= delay_link_.front()->arrival_time()) {
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// Deliver this packet.
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NetworkPacket* packet = delay_link_.front();
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delay_link_.pop();
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packet_receiver_->IncomingPacket(packet->data(), packet->data_length());
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++sent_packets_;
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// |time_now| might be later than when the packet should have arrived, due
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// to NetworkProcess being called too late. For stats, use the time it
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// should have been on the link.
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total_packet_delay_ += packet->arrival_time() - packet->send_time();
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delete packet;
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}
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}
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} // namespace webrtc
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@ -0,0 +1,205 @@
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/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <gmock/gmock.h>
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#include <gtest/gtest.h>
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/video_engine/test/libvietest/include/fake_network_pipe.h"
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using ::testing::_;
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using ::testing::AnyNumber;
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using ::testing::Return;
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using ::testing::Invoke;
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namespace webrtc {
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class MockReceiver : public PacketReceiver {
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public:
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MockReceiver() {}
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virtual ~MockReceiver() {}
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void IncomingPacket(uint8_t* data, int length) {
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IncomingData(data, length);
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delete [] data;
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}
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MOCK_METHOD2(IncomingData, void(uint8_t*, int));
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};
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class FakeNetworkPipeTest : public ::testing::Test {
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protected:
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virtual void SetUp() {
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TickTime::UseFakeClock(12345);
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receiver_.reset(new MockReceiver());
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}
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virtual void TearDown() {
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}
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void SendPackets(FakeNetworkPipe* pipe, int number_packets, int kPacketSize) {
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scoped_array<uint8_t> packet(new uint8_t[kPacketSize]);
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for (int i = 0; i < number_packets; ++i) {
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pipe->SendPacket(packet.get(), kPacketSize);
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}
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}
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int PacketTimeMs(int capacity_kbps, int kPacketSize) {
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return 8 * kPacketSize / capacity_kbps;
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}
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scoped_ptr<MockReceiver> receiver_;
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};
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void DeleteMemory(uint8_t* data, int length) { delete [] data; }
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// Test the capacity link and verify we get as many packets as we expect.
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TEST_F(FakeNetworkPipeTest, CapacityTest) {
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const int kQueueLength = 20;
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const int kNetworkDelayMs = 0;
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const int kLinkCapacityKbps = 80;
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const int kLossPercent = 0;
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scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
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kQueueLength,
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kNetworkDelayMs,
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kLinkCapacityKbps,
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kLossPercent));
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// Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
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// get through the pipe.
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const int kNumPackets = 10;
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const int kPacketSize = 1000;
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SendPackets(pipe.get(), kNumPackets , kPacketSize);
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// Time to get one packet through the link.
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const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
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// Time haven't increased yet, so we souldn't get any packets.
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(0);
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pipe->NetworkProcess();
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// Advance enough time to release one packet.
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TickTime::AdvanceFakeClock(kPacketTimeMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(1);
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pipe->NetworkProcess();
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// Release all but one packet
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TickTime::AdvanceFakeClock(9 * kPacketTimeMs - 1);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(8);
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pipe->NetworkProcess();
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// And the last one.
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TickTime::AdvanceFakeClock(1);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(1);
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pipe->NetworkProcess();
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}
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// Test the extra network delay.
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TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
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const int kQueueLength = 20;
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const int kNetworkDelayMs = 100;
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const int kLinkCapacityKbps = 80;
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const int kLossPercent = 0;
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scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
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kQueueLength,
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kNetworkDelayMs,
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kLinkCapacityKbps,
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kLossPercent));
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const int kNumPackets = 2;
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const int kPacketSize = 1000;
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SendPackets(pipe.get(), kNumPackets , kPacketSize);
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// Time to get one packet through the link.
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const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
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// Increase more than kPacketTimeMs, but not more than the extra delay.
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TickTime::AdvanceFakeClock(kPacketTimeMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(0);
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pipe->NetworkProcess();
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// Advance the network delay to get the first packet.
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TickTime::AdvanceFakeClock(kNetworkDelayMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(1);
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pipe->NetworkProcess();
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// Advance one more kPacketTimeMs to get the last packet.
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TickTime::AdvanceFakeClock(kPacketTimeMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(1);
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pipe->NetworkProcess();
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}
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// Test the number of buffers and packets are dropped when sending too many
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// packets too quickly.
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TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
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const int kQueueLength = 2;
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const int kNetworkDelayMs = 0;
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const int kLinkCapacityKbps = 80;
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const int kLossPercent = 0;
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scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
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kQueueLength,
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kNetworkDelayMs,
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kLinkCapacityKbps,
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kLossPercent));
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const int kPacketSize = 1000;
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const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
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// Send three packets and verify only 2 are delivered.
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SendPackets(pipe.get(), 3, kPacketSize);
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// Increase time enough to deliver all three packets, verify only two are
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// delivered.
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TickTime::AdvanceFakeClock(3 * kPacketTimeMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(2);
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pipe->NetworkProcess();
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}
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// Test we get statistics as expected.
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TEST_F(FakeNetworkPipeTest, StatisticsTest) {
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const int kQueueLength = 2;
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const int kNetworkDelayMs = 20;
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const int kLinkCapacityKbps = 80;
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const int kLossPercent = 0;
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scoped_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(receiver_.get(),
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kQueueLength,
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kNetworkDelayMs,
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kLinkCapacityKbps,
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kLossPercent));
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const int kPacketSize = 1000;
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const int kPacketTimeMs = PacketTimeMs(kLinkCapacityKbps, kPacketSize);
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// Send three packets and verify only 2 are delivered.
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SendPackets(pipe.get(), 3, kPacketSize);
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TickTime::AdvanceFakeClock(3 * kPacketTimeMs + kNetworkDelayMs);
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EXPECT_CALL(*receiver_, IncomingData(_, _))
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.Times(2);
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pipe->NetworkProcess();
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|
||||
// 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
|
Loading…
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Reference in New Issue
Block a user