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}
52 lines
1.4 KiB
C++
52 lines
1.4 KiB
C++
/*
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* Copyright (c) 2017 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 AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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#define AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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#include <memory>
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#include <string>
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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 "test/call_test.h"
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namespace webrtc {
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namespace test {
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class AudioBweTest : public test::EndToEndTest {
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public:
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AudioBweTest();
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protected:
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virtual std::string AudioInputFile() = 0;
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virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() = 0;
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size_t GetNumVideoStreams() const override;
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size_t GetNumAudioStreams() const override;
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size_t GetNumFlexfecStreams() const override;
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std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
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void OnFakeAudioDevicesCreated(
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TestAudioDeviceModule* send_audio_device,
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TestAudioDeviceModule* recv_audio_device) override;
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void PerformTest() override;
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private:
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TestAudioDeviceModule* send_audio_device_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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