webrtc/api/test/simulated_network.h
Jonas Olsson a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00

85 lines
2.8 KiB
C++

/*
* Copyright (c) 2018 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 API_TEST_SIMULATED_NETWORK_H_
#define API_TEST_SIMULATED_NETWORK_H_
#include <stddef.h>
#include <stdint.h>
#include <deque>
#include <queue>
#include <vector>
#include "absl/types/optional.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/random.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
struct PacketInFlightInfo {
PacketInFlightInfo(size_t size, int64_t send_time_us, uint64_t packet_id)
: size(size), send_time_us(send_time_us), packet_id(packet_id) {}
size_t size;
int64_t send_time_us;
// Unique identifier for the packet in relation to other packets in flight.
uint64_t packet_id;
};
struct PacketDeliveryInfo {
static constexpr int kNotReceived = -1;
PacketDeliveryInfo(PacketInFlightInfo source, int64_t receive_time_us)
: receive_time_us(receive_time_us), packet_id(source.packet_id) {}
int64_t receive_time_us;
uint64_t packet_id;
};
// BuiltInNetworkBehaviorConfig is a built-in network behavior configuration
// for built-in network behavior that will be used by WebRTC if no custom
// NetworkBehaviorInterface is provided.
struct BuiltInNetworkBehaviorConfig {
BuiltInNetworkBehaviorConfig() {}
// Queue length in number of packets.
size_t queue_length_packets = 0;
// Delay in addition to capacity induced delay.
int queue_delay_ms = 0;
// Standard deviation of the extra delay.
int delay_standard_deviation_ms = 0;
// Link capacity in kbps.
int link_capacity_kbps = 0;
// Random packet loss.
int loss_percent = 0;
// If packets are allowed to be reordered.
bool allow_reordering = false;
// The average length of a burst of lost packets.
int avg_burst_loss_length = -1;
// Additional bytes to add to packet size.
int packet_overhead = 0;
// Enable CoDel active queue management.
bool codel_active_queue_management = false;
};
class NetworkBehaviorInterface {
public:
virtual bool EnqueuePacket(PacketInFlightInfo packet_info) = 0;
// Retrieves all packets that should be delivered by the given receive time.
virtual std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
int64_t receive_time_us) = 0;
// Returns time in microseconds when caller should call
// DequeueDeliverablePackets to get next set of packets to deliver.
virtual absl::optional<int64_t> NextDeliveryTimeUs() const = 0;
virtual ~NetworkBehaviorInterface() = default;
};
} // namespace webrtc
#endif // API_TEST_SIMULATED_NETWORK_H_