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836fee1e1a
Currently there's an implicit requirement that users of SimulatedNetwork should call it repeatedly, even if the return value of NextDeliveryTimeUs is unset. With this change, it will indicate that there might be a delivery in 5 ms at any time there are packets in queue. Which results in unchanged behavior compared to current usage but allows new users to expect robust behavior. Bug: webrtc:9510 Change-Id: I45b8b5f1f0d3d13a8ec9b163d4011c5f01a53069 Reviewed-on: https://webrtc-review.googlesource.com/c/120402 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26617}
83 lines
2.6 KiB
C++
83 lines
2.6 KiB
C++
/*
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* Copyright (c) 2013 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 TEST_DIRECT_TRANSPORT_H_
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#define TEST_DIRECT_TRANSPORT_H_
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#include <memory>
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#include "api/call/transport.h"
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#include "api/test/simulated_network.h"
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#include "call/call.h"
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#include "call/simulated_packet_receiver.h"
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#include "rtc_base/sequenced_task_checker.h"
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#include "rtc_base/thread_annotations.h"
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#include "test/single_threaded_task_queue.h"
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namespace webrtc {
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class Clock;
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class PacketReceiver;
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namespace test {
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class Demuxer {
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public:
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explicit Demuxer(const std::map<uint8_t, MediaType>& payload_type_map);
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~Demuxer() = default;
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MediaType GetMediaType(const uint8_t* packet_data,
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const size_t packet_length) const;
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const std::map<uint8_t, MediaType> payload_type_map_;
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RTC_DISALLOW_COPY_AND_ASSIGN(Demuxer);
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};
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// Objects of this class are expected to be allocated and destroyed on the
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// same task-queue - the one that's passed in via the constructor.
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class DirectTransport : public Transport {
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public:
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DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
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std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
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Call* send_call,
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const std::map<uint8_t, MediaType>& payload_type_map);
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~DirectTransport() override;
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RTC_DEPRECATED void StopSending();
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// TODO(holmer): Look into moving this to the constructor.
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virtual void SetReceiver(PacketReceiver* receiver);
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bool SendRtp(const uint8_t* data,
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size_t length,
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const PacketOptions& options) override;
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bool SendRtcp(const uint8_t* data, size_t length) override;
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int GetAverageDelayMs();
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private:
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void ProcessPackets() RTC_EXCLUSIVE_LOCKS_REQUIRED(&process_lock_);
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void SendPacket(const uint8_t* data, size_t length);
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void Start();
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Call* const send_call_;
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Clock* const clock_;
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SingleThreadedTaskQueueForTesting* const task_queue_;
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rtc::CriticalSection process_lock_;
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absl::optional<SingleThreadedTaskQueueForTesting::TaskId> next_process_task_
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RTC_GUARDED_BY(&process_lock_);
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const Demuxer demuxer_;
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const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_DIRECT_TRANSPORT_H_
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