Index: webrtc/audio/test/audio_bwe_integration_test.cc |
diff --git a/webrtc/audio/test/audio_bwe_integration_test.cc b/webrtc/audio/test/audio_bwe_integration_test.cc |
deleted file mode 100644 |
index bb5d9165436382b42fcbf0c6da7f705643d14c06..0000000000000000000000000000000000000000 |
--- a/webrtc/audio/test/audio_bwe_integration_test.cc |
+++ /dev/null |
@@ -1,146 +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 "webrtc/audio/test/audio_bwe_integration_test.h" |
- |
-#include "webrtc/base/ptr_util.h" |
-#include "webrtc/common_audio/wav_file.h" |
-#include "webrtc/system_wrappers/include/sleep.h" |
-#include "webrtc/test/field_trial.h" |
-#include "webrtc/test/gtest.h" |
-#include "webrtc/test/testsupport/fileutils.h" |
- |
-namespace webrtc { |
-namespace test { |
- |
-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<test::FakeAudioDevice::Capturer> |
-AudioBweTest::CreateCapturer() { |
- return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile()); |
-} |
- |
-void AudioBweTest::OnFakeAudioDevicesCreated( |
- test::FakeAudioDevice* send_audio_device, |
- test::FakeAudioDevice* recv_audio_device) { |
- send_audio_device_ = send_audio_device; |
-} |
- |
-test::PacketTransport* AudioBweTest::CreateSendTransport(Call* sender_call) { |
- return new test::PacketTransport( |
- sender_call, this, test::PacketTransport::kSender, |
- test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
-} |
- |
-test::PacketTransport* AudioBweTest::CreateReceiveTransport() { |
- return new test::PacketTransport( |
- nullptr, this, test::PacketTransport::kReceiver, |
- test::CallTest::payload_type_map_, GetNetworkPipeConfig()); |
-} |
- |
-void AudioBweTest::PerformTest() { |
- send_audio_device_->WaitForRecordingEnd(); |
- SleepMs(GetNetworkPipeConfig().queue_delay_ms); |
-} |
- |
-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, 30000); |
- 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 = |
- rtc::Optional<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; |
- |
-TEST_F(AudioBweIntegrationTest, NoBandwidthDropAfterDtx) { |
- webrtc::test::ScopedFieldTrials override_field_trials( |
- "WebRTC-Audio-SendSideBwe/Enabled/" |
- "WebRTC-SendSideBwe-WithOverhead/Enabled/"); |
- NoBandwidthDropAfterDtx test; |
- RunBaseTest(&test); |
-} |
- |
-} // namespace test |
-} // namespace webrtc |