blob: ecdd1626203575612d260aed4796346e731c4bca [file] [log] [blame]
/*
* 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 "common_audio/wav_file.h"
#include "rtc_base/ptr_util.h"
#include "system_wrappers/include/sleep.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
namespace webrtc {
namespace test {
namespace {
// Wait a second between stopping sending and stopping receiving audio.
constexpr int kExtraProcessTimeMs = 1000;
} // namespace
AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
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;
}
test::PacketTransport* AudioBweTest::CreateSendTransport(
SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) {
return new test::PacketTransport(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}
test::PacketTransport* AudioBweTest::CreateReceiveTransport(
SingleThreadedTaskQueueForTesting* task_queue) {
return new test::PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}
void AudioBweTest::PerformTest() {
send_audio_device_->WaitForRecordingEnd();
SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraProcessTimeMs);
}
class StatsPollTask : public rtc::QueuedTask {
public:
explicit StatsPollTask(Call* sender_call) : sender_call_(sender_call) {}
private:
bool Run() override {
RTC_CHECK(sender_call_);
Call::Stats call_stats = sender_call_->GetStats();
EXPECT_GT(call_stats.send_bandwidth_bps, 25000);
rtc::TaskQueue::Current()->PostDelayedTask(
std::unique_ptr<QueuedTask>(this), 100);
return false;
}
Call* sender_call_;
};
class NoBandwidthDropAfterDtx : public AudioBweTest {
public:
NoBandwidthDropAfterDtx()
: sender_call_(nullptr), stats_poller_("stats poller task queue") {}
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::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 (AudioReceiveStream::Config& recv_config : *receive_configs) {
recv_config.rtp.transport_cc = true;
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");
}
FakeNetworkPipe::Config GetNetworkPipeConfig() override {
FakeNetworkPipe::Config 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(
std::unique_ptr<rtc::QueuedTask>(new StatsPollTask(sender_call_)), 100);
sender_call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO, 0);
AudioBweTest::PerformTest();
}
private:
Call* sender_call_;
rtc::TaskQueue 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) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/");
NoBandwidthDropAfterDtx test;
RunBaseTest(&test);
}
} // namespace test
} // namespace webrtc