Test and fix for huge bwe drop after alr state.

webrtc/audio/test/low_bandwidth_audio_test.cc

 /*
  *  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 <algorithm>
 
 #include "gflags/gflags.h"
+
 #include "webrtc/audio/test/low_bandwidth_audio_test.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/common_audio/wav_file.h"
+#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/field_trial.h"
+#include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
-#include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
+using ::testing::NiceMock;
+using ::testing::Invoke;
+using ::testing::_;
 
 DEFINE_int32(sample_rate_hz, 16000,
              "Sample rate (Hz) of the produced audio files.");
 
 DEFINE_bool(quick, false,
             "Don't do the full audio recording. "
             "Used to quickly check that the test runs without crashing.");
 
 namespace {
-
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
 std::string FileSampleRateSuffix() {
   return std::to_string(FLAGS_sample_rate_hz / 1000);
 }
 
 }  // namespace
 
 namespace webrtc {
@@ skipping 40 matching lines @@
     test::FakeAudioDevice* recv_audio_device) {
   send_audio_device_ = send_audio_device;
 }
 
 FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
   return FakeNetworkPipe::Config();
 }
 
 test::PacketTransport* AudioQualityTest::CreateSendTransport(
     Call* sender_call) {
+  sender_call_ = sender_call;
   return new test::PacketTransport(
       sender_call, this, test::PacketTransport::kSender,
       test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
   return new test::PacketTransport(
       nullptr, this, test::PacketTransport::kReceiver,
       test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
   // Large bitrate by default.
   const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
                                               {{"stereo", "1"}});
   send_config->send_codec_spec =
       rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
           {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
 }
 
 void AudioQualityTest::PerformTest() {
+  sender_call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO, 0);
+
   if (FLAGS_quick) {
     // Let the recording run for a small amount of time to check if it works.
     SleepMs(1000);
   } else {
     // Wait until the input audio file is done...
     send_audio_device_->WaitForRecordingEnd();
     // and some extra time to account for network delay.
     SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
   }
 }
@@ skipping 38 matching lines @@
     pipe_config.queue_delay_ms = 400;
     return pipe_config;
   }
 };
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
 
+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 AudioQualityTest {
+ public:
+  NoBandwidthDropAfterDtx() : 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 PerformTest() override {
+    stats_poller_.PostDelayedTask(
+        std::unique_ptr<rtc::QueuedTask>(new StatsPollTask(sender_call_)), 100);
+    AudioQualityTest::PerformTest();
+  }
+
+ private:
+  rtc::TaskQueue stats_poller_;
+};
+
+TEST_F(LowBandwidthAudioTest, NoBandwidthDropAfterDtx) {
+  webrtc::test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-SendSideBwe/Enabled/"
+      "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+  NoBandwidthDropAfterDtx test;
+  RunBaseTest(&test);
+}
+
 }  // namespace test
 }  // namespace webrtc