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Delete unused Audio Bwe integration test.

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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}
This commit is contained in:
Per K 2023-01-25 13:43:01 +01:00 committed by WebRTC LUCI CQ
parent cfbb247f6d
commit 73e0cc8969
4 changed files with 0 additions and 220 deletions
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1
BUILD.gn
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@ -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",

28
audio/BUILD.gn
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@ -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" ]
}
}
}

140
audio/test/audio_bwe_integration_test.cc
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@ -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

51
audio/test/audio_bwe_integration_test.h
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@ -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_