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e87cfe2315
Add method FrameType for frame to TestConfig. Bug: none Change-Id: Icfeb12fcb961559c9b36a3aedb081a840b9d8556 Reviewed-on: https://webrtc-review.googlesource.com/16120 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20458}
112 lines
4 KiB
C++
112 lines
4 KiB
C++
/*
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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 MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
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#define MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
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#include <stdlib.h>
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#include "modules/video_coding/include/video_codec_interface.h"
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#include "rtc_base/criticalsection.h"
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#include "test/testsupport/packet_reader.h"
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namespace webrtc {
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namespace test {
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// Which mode the packet loss shall be performed according to.
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enum PacketLossMode {
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// Drops packets with a configured probability independently for each packet
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kUniform,
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// Drops packets similar to uniform but when a packet is being dropped, the
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// number of lost packets in a row is equal to the configured burst length.
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kBurst
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};
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// Contains configurations related to networking and simulation of
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// scenarios caused by network interference.
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struct NetworkingConfig {
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NetworkingConfig()
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: packet_size_in_bytes(1500),
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max_payload_size_in_bytes(1440),
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packet_loss_mode(kUniform),
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packet_loss_probability(0.0),
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packet_loss_burst_length(1) {}
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// Packet size in bytes. Default: 1500 bytes.
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size_t packet_size_in_bytes;
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// Encoder specific setting of maximum size in bytes of each payload.
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// Default: 1440 bytes.
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size_t max_payload_size_in_bytes;
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// Packet loss mode. Two different packet loss models are supported:
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// uniform or burst. This setting has no effect unless
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// packet_loss_probability is >0.
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// Default: uniform.
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PacketLossMode packet_loss_mode;
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// Packet loss probability. A value between 0.0 and 1.0 that defines the
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// probability of a packet being lost. 0.1 means 10% and so on.
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// Default: 0 (no loss).
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double packet_loss_probability;
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// Packet loss burst length. Defines how many packets will be lost in a burst
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// when a packet has been decided to be lost. Must be >=1. Default: 1.
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int packet_loss_burst_length;
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};
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// Class for simulating packet loss on the encoded frame data.
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// When a packet loss has occurred in a frame, the remaining data in that
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// frame is lost (even if burst length is only a single packet).
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// TODO(kjellander): Support discarding only individual packets in the frame
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// when CL 172001 has been submitted. This also requires a correct
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// fragmentation header to be passed to the decoder.
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//
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// To get a repeatable packet drop pattern, re-initialize the random seed
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// using InitializeRandomSeed before each test run.
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class PacketManipulator {
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public:
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virtual ~PacketManipulator() {}
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// Manipulates the data of the encoded_image to simulate parts being lost
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// during transport.
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// If packets are dropped from frame data, the completedFrame field will be
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// set to false.
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// Returns the number of packets being dropped.
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virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
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};
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class PacketManipulatorImpl : public PacketManipulator {
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public:
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PacketManipulatorImpl(PacketReader* packet_reader,
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const NetworkingConfig& config,
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bool verbose);
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~PacketManipulatorImpl() = default;
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int ManipulatePackets(webrtc::EncodedImage* encoded_image) override;
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virtual void InitializeRandomSeed(unsigned int seed);
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protected:
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// Returns a uniformly distributed random value between 0.0 and 1.0
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virtual double RandomUniform();
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private:
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PacketReader* const packet_reader_;
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const NetworkingConfig& config_;
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const bool verbose_;
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// Used to simulate a burst over several frames.
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int active_burst_packets_;
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rtc::CriticalSection critsect_;
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unsigned int random_seed_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
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