From 73e0cc8969510b224f17cf8afa3abcd874143640 Mon Sep 17 00:00:00 2001 From: Per K Date: Wed, 25 Jan 2023 13:43:01 +0100 Subject: [PATCH] Delete unused Audio Bwe integration test. MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Bug: none Change-Id: Id8eb4ad4630820441d18e8d1449f4a8d87da9a59 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291335 Reviewed-by: Jakob Ivarsson‎ Commit-Queue: Per Kjellander Reviewed-by: Mirko Bonadei Cr-Commit-Position: refs/heads/main@{#39202} --- BUILD.gn | 1 - audio/BUILD.gn | 28 ----- audio/test/audio_bwe_integration_test.cc | 140 ----------------------- audio/test/audio_bwe_integration_test.h | 51 --------- 4 files changed, 220 deletions(-) delete mode 100644 audio/test/audio_bwe_integration_test.cc delete mode 100644 audio/test/audio_bwe_integration_test.h diff --git a/BUILD.gn b/BUILD.gn index c41059f288..5817d22227 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -700,7 +700,6 @@ if (rtc_include_tests && !build_with_chromium) { rtc_test("webrtc_perf_tests") { testonly = true deps = [ - "audio:audio_perf_tests", "call:call_perf_tests", "modules/audio_coding:audio_coding_perf_tests", "modules/audio_processing:audio_processing_perf_tests", diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 183a99b1a1..919140e15c 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -331,32 +331,4 @@ if (rtc_include_tests) { } } } - - if (!build_with_chromium) { - rtc_library("audio_perf_tests") { - testonly = true - - sources = [ - "test/audio_bwe_integration_test.cc", - "test/audio_bwe_integration_test.h", - ] - deps = [ - "../api:simulated_network_api", - "../api/task_queue", - "../call:fake_network", - "../call:simulated_network", - "../common_audio", - "../rtc_base:task_queue_for_test", - "../system_wrappers", - "../test:field_trial", - "../test:fileutils", - "../test:test_common", - "../test:test_main", - "../test:test_support", - "//testing/gtest", - ] - absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ] - data = [ "//resources/voice_engine/audio_dtx16.wav" ] - } - } } diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc deleted file mode 100644 index fd68cbd627..0000000000 --- a/audio/test/audio_bwe_integration_test.cc +++ /dev/null @@ -1,140 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "audio/test/audio_bwe_integration_test.h" - -#include - -#include "absl/functional/any_invocable.h" -#include "api/task_queue/task_queue_base.h" -#include "call/fake_network_pipe.h" -#include "call/simulated_network.h" -#include "common_audio/wav_file.h" -#include "rtc_base/task_queue_for_test.h" -#include "system_wrappers/include/sleep.h" -#include "test/field_trial.h" -#include "test/gtest.h" -#include "test/testsupport/file_utils.h" - -namespace webrtc { -namespace test { - -namespace { -enum : int { // The first valid value is 1. - kTransportSequenceNumberExtensionId = 1, -}; - -// Wait a second between stopping sending and stopping receiving audio. -constexpr int kExtraProcessTimeMs = 1000; -} // namespace - -AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeout) {} - -size_t AudioBweTest::GetNumVideoStreams() const { - return 0; -} -size_t AudioBweTest::GetNumAudioStreams() const { - return 1; -} -size_t AudioBweTest::GetNumFlexfecStreams() const { - return 0; -} - -std::unique_ptr -AudioBweTest::CreateCapturer() { - return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile()); -} - -void AudioBweTest::OnFakeAudioDevicesCreated( - TestAudioDeviceModule* send_audio_device, - TestAudioDeviceModule* recv_audio_device) { - send_audio_device_ = send_audio_device; -} - -void AudioBweTest::PerformTest() { - send_audio_device_->WaitForRecordingEnd(); - SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraProcessTimeMs); -} - -absl::AnyInvocable StatsPollTask(Call* sender_call) { - RTC_CHECK(sender_call); - return [sender_call] { - Call::Stats call_stats = sender_call->GetStats(); - EXPECT_GT(call_stats.send_bandwidth_bps, 25000); - TaskQueueBase::Current()->PostDelayedTask(StatsPollTask(sender_call), - TimeDelta::Millis(100)); - }; -} - -class NoBandwidthDropAfterDtx : public AudioBweTest { - public: - NoBandwidthDropAfterDtx() - : sender_call_(nullptr), stats_poller_("stats poller task queue") {} - - void ModifyAudioConfigs(AudioSendStream::Config* send_config, - std::vector* - receive_configs) override { - send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( - test::CallTest::kAudioSendPayloadType, - {"OPUS", - 48000, - 2, - {{"ptime", "60"}, {"usedtx", "1"}, {"stereo", "1"}}}); - - send_config->min_bitrate_bps = 6000; - send_config->max_bitrate_bps = 100000; - send_config->rtp.extensions.push_back( - RtpExtension(RtpExtension::kTransportSequenceNumberUri, - kTransportSequenceNumberExtensionId)); - for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) { - recv_config.rtp.extensions = send_config->rtp.extensions; - recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; - } - } - - std::string AudioInputFile() override { - return test::ResourcePath("voice_engine/audio_dtx16", "wav"); - } - - BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() override { - BuiltInNetworkBehaviorConfig pipe_config; - pipe_config.link_capacity_kbps = 50; - pipe_config.queue_length_packets = 1500; - pipe_config.queue_delay_ms = 300; - return pipe_config; - } - - void OnCallsCreated(Call* sender_call, Call* receiver_call) override { - sender_call_ = sender_call; - } - - void PerformTest() override { - stats_poller_.PostDelayedTask(StatsPollTask(sender_call_), - TimeDelta::Millis(100)); - sender_call_->OnAudioTransportOverheadChanged(0); - AudioBweTest::PerformTest(); - } - - private: - Call* sender_call_; - TaskQueueForTest stats_poller_; -}; - -using AudioBweIntegrationTest = CallTest; - -// TODO(tschumim): This test is flaky when run on android and mac. Re-enable the -// test for when the issue is fixed. -TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) { - NoBandwidthDropAfterDtx test; - RunBaseTest(&test); -} - -} // namespace test -} // namespace webrtc diff --git a/audio/test/audio_bwe_integration_test.h b/audio/test/audio_bwe_integration_test.h deleted file mode 100644 index 132ab0a3a5..0000000000 --- a/audio/test/audio_bwe_integration_test.h +++ /dev/null @@ -1,51 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#ifndef AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_ -#define AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_ - -#include -#include - -#include "api/task_queue/task_queue_base.h" -#include "api/test/simulated_network.h" -#include "test/call_test.h" - -namespace webrtc { -namespace test { - -class AudioBweTest : public test::EndToEndTest { - public: - AudioBweTest(); - - protected: - virtual std::string AudioInputFile() = 0; - - virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() = 0; - - size_t GetNumVideoStreams() const override; - size_t GetNumAudioStreams() const override; - size_t GetNumFlexfecStreams() const override; - - std::unique_ptr CreateCapturer() override; - - void OnFakeAudioDevicesCreated( - TestAudioDeviceModule* send_audio_device, - TestAudioDeviceModule* recv_audio_device) override; - - void PerformTest() override; - - private: - TestAudioDeviceModule* send_audio_device_; -}; - -} // namespace test -} // namespace webrtc - -#endif // AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_