89870ffa95
This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
96 lines
3.1 KiB
C++
96 lines
3.1 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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// TODO(perkj, https://bugs.webrtc.org/7135): Remove header once downstream
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// projects have been updated.
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[[deprecated("Use ctor that provide header extensions.")]] DirectTransport(
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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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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 bool use_legacy_send_;
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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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