Reimplemeted "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
new file mode 100644
index 0000000000000000000000000000000000000000..bb5d9165436382b42fcbf0c6da7f705643d14c06
--- /dev/null
+++ b/webrtc/audio/test/audio_bwe_integration_test.cc
@@ -0,0 +1,146 @@
+/*
+ *  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);

[stefan-webrtc, 2017/07/06 15:28:35]

+1000 to ensure that data in the receiver's queue also has time to get
processed?

[tschumi, 2017/07/07 07:51:14]

Ok

+}
+
+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