Delete unused Audio Bwe integration test.
Bug: none
Change-Id: Id8eb4ad4630820441d18e8d1449f4a8d87da9a59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291335
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39202}
diff --git a/BUILD.gn b/BUILD.gn
index c41059f..5817d22 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -700,7 +700,6 @@
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 183a99b..919140e 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -331,32 +331,4 @@
}
}
}
-
- 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 fd68cbd..0000000
--- 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 <memory>
-
-#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<TestAudioDeviceModule::Capturer>
-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<void() &&> 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<AudioReceiveStreamInterface::Config>*
- 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 132ab0a..0000000
--- 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 <memory>
-#include <string>
-
-#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<TestAudioDeviceModule::Capturer> 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_
