Revert of Test and fix for huge bwe drop after alr state.

webrtc/audio/test/audio_bwe_integration_test.cc

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