664cf14f9f
This reverts commit f2a083f262d86737893e774c696716742fcab3e3. Reason for revert: Test problem fixed in https://webrtc-review.googlesource.com/c/src/+/291333. Original change's description: > Revert "Delete PacketReceiver::DeliverPacket from all implementations" > > This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63. > > Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200 > > Original change's description: > > Delete PacketReceiver::DeliverPacket from all implementations > > > > And fix tests that still depend on extensions to be known by the receiver. > > > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > > > Bug: webrtc:7135,webrtc:14795 > > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996 > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39184} > > Bug: webrtc:7135,webrtc:14795,b/266658815 > Change-Id: I9d03f4952938d176ffee110a707acadc1846457c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400 > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#39189} Bug: webrtc:7135,webrtc:14795,b/266658815 Change-Id: Ia640f4342a1f42012ba5295003e17aef7613ad80 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291440 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39199}
87 lines
2.7 KiB
C++
87 lines
2.7 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/sequence_checker.h"
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#include "api/task_queue/task_queue_base.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/synchronization/mutex.h"
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#include "rtc_base/task_utils/repeating_task.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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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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Demuxer(const Demuxer&) = delete;
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Demuxer& operator=(const Demuxer&) = delete;
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MediaType GetMediaType(const uint8_t* packet_data,
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size_t packet_length) const;
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const std::map<uint8_t, MediaType> payload_type_map_;
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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(TaskQueueBase* 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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rtc::ArrayView<const RtpExtension> audio_extensions,
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rtc::ArrayView<const RtpExtension> video_extensions);
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~DirectTransport() override;
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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 LegacySendPacket(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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TaskQueueBase* const task_queue_;
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Mutex process_lock_;
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RepeatingTaskHandle next_process_task_ 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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const RtpHeaderExtensionMap audio_extensions_;
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const RtpHeaderExtensionMap video_extensions_;
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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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