Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 949 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
new file mode 100644
index 0000000000..a81b40cbe7
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
@@ -0,0 +1,949 @@
+/*
+ *  Copyright (c) 2015 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/audio_send_stream.h"
+
+#include <memory>
+#include <string>
+#include <thread>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/test/mock_frame_encryptor.h"
+#include "audio/audio_state.h"
+#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
+#include "call/test/mock_rtp_transport_controller_send.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "modules/utility/maybe_worker_thread.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/mock_audio_encoder_factory.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/real_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::Ne;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::StrEq;
+
+static const float kTolerance = 0.0001f;
+
+const uint32_t kSsrc = 1234;
+const char* kCName = "foo_name";
+const int kAudioLevelId = 2;
+const int kTransportSequenceNumberId = 4;
+const int32_t kEchoDelayMedian = 254;
+const int32_t kEchoDelayStdDev = -3;
+const double kDivergentFilterFraction = 0.2f;
+const double kEchoReturnLoss = -65;
+const double kEchoReturnLossEnhancement = 101;
+const double kResidualEchoLikelihood = -1.0f;
+const double kResidualEchoLikelihoodMax = 23.0f;
+const CallSendStatistics kCallStats = {112, 12, 13456, 17890};
+const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
+const int kTelephoneEventPayloadType = 123;
+const int kTelephoneEventPayloadFrequency = 65432;
+const int kTelephoneEventCode = 45;
+const int kTelephoneEventDuration = 6789;
+constexpr int kIsacPayloadType = 103;
+const SdpAudioFormat kIsacFormat = {"isac", 16000, 1};
+const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
+const SdpAudioFormat kG722Format = {"g722", 8000, 1};
+const AudioCodecSpec kCodecSpecs[] = {
+    {kIsacFormat, {16000, 1, 32000, 10000, 32000}},
+    {kOpusFormat, {48000, 1, 32000, 6000, 510000}},
+    {kG722Format, {16000, 1, 64000}}};
+
+// TODO(dklee): This mirrors calculation in audio_send_stream.cc, which
+// should be made more precise in the future. This can be changed when that
+// logic is more accurate.
+const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
+const TimeDelta kMinFrameLength = TimeDelta::Millis(20);
+const TimeDelta kMaxFrameLength = TimeDelta::Millis(120);
+const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength;
+const DataRate kMaxOverheadRate = kOverheadPerPacket / kMinFrameLength;
+
+class MockLimitObserver : public BitrateAllocator::LimitObserver {
+ public:
+  MOCK_METHOD(void,
+              OnAllocationLimitsChanged,
+              (BitrateAllocationLimits),
+              (override));
+};
+
+std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
+    int payload_type,
+    const SdpAudioFormat& format) {
+  for (const auto& spec : kCodecSpecs) {
+    if (format == spec.format) {
+      std::unique_ptr<MockAudioEncoder> encoder(
+          new ::testing::NiceMock<MockAudioEncoder>());
+      ON_CALL(*encoder.get(), SampleRateHz())
+          .WillByDefault(Return(spec.info.sample_rate_hz));
+      ON_CALL(*encoder.get(), NumChannels())
+          .WillByDefault(Return(spec.info.num_channels));
+      ON_CALL(*encoder.get(), RtpTimestampRateHz())
+          .WillByDefault(Return(spec.format.clockrate_hz));
+      ON_CALL(*encoder.get(), GetFrameLengthRange())
+          .WillByDefault(Return(absl::optional<std::pair<TimeDelta, TimeDelta>>{
+              {TimeDelta::Millis(20), TimeDelta::Millis(120)}}));
+      return encoder;
+    }
+  }
+  return nullptr;
+}
+
+rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
+  rtc::scoped_refptr<MockAudioEncoderFactory> factory =
+      rtc::make_ref_counted<MockAudioEncoderFactory>();
+  ON_CALL(*factory.get(), GetSupportedEncoders())
+      .WillByDefault(Return(std::vector<AudioCodecSpec>(
+          std::begin(kCodecSpecs), std::end(kCodecSpecs))));
+  ON_CALL(*factory.get(), QueryAudioEncoder(_))
+      .WillByDefault(Invoke(
+          [](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
+            for (const auto& spec : kCodecSpecs) {
+              if (format == spec.format) {
+                return spec.info;
+              }
+            }
+            return absl::nullopt;
+          }));
+  ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
+      .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
+                               absl::optional<AudioCodecPairId> codec_pair_id,
+                               std::unique_ptr<AudioEncoder>* return_value) {
+        *return_value = SetupAudioEncoderMock(payload_type, format);
+      }));
+  return factory;
+}
+
+struct ConfigHelper {
+  ConfigHelper(bool audio_bwe_enabled,
+               bool expect_set_encoder_call,
+               bool use_null_audio_processing)
+      : stream_config_(/*send_transport=*/nullptr),
+        audio_processing_(
+            use_null_audio_processing
+                ? nullptr
+                : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>()),
+        bitrate_allocator_(&limit_observer_),
+        worker_queue_(field_trials,
+                      "ConfigHelper_worker_queue",
+                      time_controller_.GetTaskQueueFactory()),
+        audio_encoder_(nullptr) {
+    using ::testing::Invoke;
+
+    AudioState::Config config;
+    config.audio_mixer = AudioMixerImpl::Create();
+    config.audio_processing = audio_processing_;
+    config.audio_device_module = rtc::make_ref_counted<MockAudioDeviceModule>();
+    audio_state_ = AudioState::Create(config);
+
+    SetupDefaultChannelSend(audio_bwe_enabled);
+    SetupMockForSetupSendCodec(expect_set_encoder_call);
+    SetupMockForCallEncoder();
+
+    // Use ISAC as default codec so as to prevent unnecessary `channel_proxy_`
+    // calls from the default ctor behavior.
+    stream_config_.send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
+    stream_config_.rtp.ssrc = kSsrc;
+    stream_config_.rtp.c_name = kCName;
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+    if (audio_bwe_enabled) {
+      AddBweToConfig(&stream_config_);
+    }
+    stream_config_.encoder_factory = SetupEncoderFactoryMock();
+    stream_config_.min_bitrate_bps = 10000;
+    stream_config_.max_bitrate_bps = 65000;
+  }
+
+  std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
+    EXPECT_CALL(rtp_transport_, GetWorkerQueue())
+        .WillRepeatedly(Return(&worker_queue_));
+    return std::unique_ptr<internal::AudioSendStream>(
+        new internal::AudioSendStream(
+            time_controller_.GetClock(), stream_config_, audio_state_,
+            time_controller_.GetTaskQueueFactory(), &rtp_transport_,
+            &bitrate_allocator_, &event_log_, absl::nullopt,
+            std::unique_ptr<voe::ChannelSendInterface>(channel_send_),
+            field_trials));
+  }
+
+  AudioSendStream::Config& config() { return stream_config_; }
+  MockAudioEncoderFactory& mock_encoder_factory() {
+    return *static_cast<MockAudioEncoderFactory*>(
+        stream_config_.encoder_factory.get());
+  }
+  MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; }
+  MockChannelSend* channel_send() { return channel_send_; }
+  RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
+
+  static void AddBweToConfig(AudioSendStream::Config* config) {
+    config->rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
+    config->send_codec_spec->transport_cc_enabled = true;
+  }
+
+  void SetupDefaultChannelSend(bool audio_bwe_enabled) {
+    EXPECT_TRUE(channel_send_ == nullptr);
+    channel_send_ = new ::testing::StrictMock<MockChannelSend>();
+    EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
+      return &this->rtp_rtcp_;
+    }));
+    EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc));
+    EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+    EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
+    EXPECT_CALL(*channel_send_, SetEncoderToPacketizerFrameTransformer(_))
+        .Times(1);
+    EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1);
+    EXPECT_CALL(*channel_send_,
+                SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
+        .Times(1);
+    EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
+        .WillRepeatedly(Return(&bandwidth_observer_));
+    if (audio_bwe_enabled) {
+      EXPECT_CALL(rtp_rtcp_,
+                  RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
+                                             kTransportSequenceNumberId))
+          .Times(1);
+      EXPECT_CALL(*channel_send_,
+                  RegisterSenderCongestionControlObjects(
+                      &rtp_transport_, Eq(&bandwidth_observer_)))
+          .Times(1);
+    } else {
+      EXPECT_CALL(*channel_send_, RegisterSenderCongestionControlObjects(
+                                      &rtp_transport_, Eq(nullptr)))
+          .Times(1);
+    }
+    EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
+  }
+
+  void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
+    if (expect_set_encoder_call) {
+      EXPECT_CALL(*channel_send_, SetEncoder)
+          .WillOnce(
+              [this](int payload_type, std::unique_ptr<AudioEncoder> encoder) {
+                this->audio_encoder_ = std::move(encoder);
+                return true;
+              });
+    }
+  }
+
+  void SetupMockForCallEncoder() {
+    // Let ModifyEncoder to invoke mock audio encoder.
+    EXPECT_CALL(*channel_send_, CallEncoder(_))
+        .WillRepeatedly(
+            [this](rtc::FunctionView<void(AudioEncoder*)> modifier) {
+              if (this->audio_encoder_)
+                modifier(this->audio_encoder_.get());
+            });
+  }
+
+  void SetupMockForSendTelephoneEvent() {
+    EXPECT_TRUE(channel_send_);
+    EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType(
+                                    kTelephoneEventPayloadType,
+                                    kTelephoneEventPayloadFrequency));
+    EXPECT_CALL(
+        *channel_send_,
+        SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
+        .WillOnce(Return(true));
+  }
+
+  void SetupMockForGetStats(bool use_null_audio_processing) {
+    using ::testing::DoAll;
+    using ::testing::SetArgPointee;
+    using ::testing::SetArgReferee;
+
+    std::vector<ReportBlock> report_blocks;
+    webrtc::ReportBlock block = kReportBlock;
+    report_blocks.push_back(block);  // Has wrong SSRC.
+    block.source_SSRC = kSsrc;
+    report_blocks.push_back(block);  // Correct block.
+    block.fraction_lost = 0;
+    report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
+
+    EXPECT_TRUE(channel_send_);
+    EXPECT_CALL(*channel_send_, GetRTCPStatistics())
+        .WillRepeatedly(Return(kCallStats));
+    EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks())
+        .WillRepeatedly(Return(report_blocks));
+    EXPECT_CALL(*channel_send_, GetANAStatistics())
+        .WillRepeatedly(Return(ANAStats()));
+    EXPECT_CALL(*channel_send_, GetTargetBitrate()).WillRepeatedly(Return(0));
+
+    audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
+    audio_processing_stats_.echo_return_loss_enhancement =
+        kEchoReturnLossEnhancement;
+    audio_processing_stats_.delay_median_ms = kEchoDelayMedian;
+    audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev;
+    audio_processing_stats_.divergent_filter_fraction =
+        kDivergentFilterFraction;
+    audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood;
+    audio_processing_stats_.residual_echo_likelihood_recent_max =
+        kResidualEchoLikelihoodMax;
+    if (!use_null_audio_processing) {
+      ASSERT_TRUE(audio_processing_);
+      EXPECT_CALL(*audio_processing_, GetStatistics(true))
+          .WillRepeatedly(Return(audio_processing_stats_));
+    }
+  }
+
+  MaybeWorkerThread* worker() { return &worker_queue_; }
+
+  test::ScopedKeyValueConfig field_trials;
+
+ private:
+  RealTimeController time_controller_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  AudioSendStream::Config stream_config_;
+  ::testing::StrictMock<MockChannelSend>* channel_send_ = nullptr;
+  rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
+  AudioProcessingStats audio_processing_stats_;
+  ::testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
+  ::testing::NiceMock<MockRtcEventLog> event_log_;
+  ::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
+  ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
+  ::testing::NiceMock<MockLimitObserver> limit_observer_;
+  BitrateAllocator bitrate_allocator_;
+  // `worker_queue` is defined last to ensure all pending tasks are cancelled
+  // and deleted before any other members.
+  MaybeWorkerThread worker_queue_;
+  std::unique_ptr<AudioEncoder> audio_encoder_;
+};
+
+// The audio level ranges linearly [0,32767].
+std::unique_ptr<AudioFrame> CreateAudioFrame1kHzSineWave(int16_t audio_level,
+                                                         int duration_ms,
+                                                         int sample_rate_hz,
+                                                         size_t num_channels) {
+  size_t samples_per_channel = sample_rate_hz / (1000 / duration_ms);
+  std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
+  std::unique_ptr<AudioFrame> audio_frame = std::make_unique<AudioFrame>();
+  audio_frame->UpdateFrame(0 /* RTP timestamp */, &audio_data[0],
+                           samples_per_channel, sample_rate_hz,
+                           AudioFrame::SpeechType::kNormalSpeech,
+                           AudioFrame::VADActivity::kVadUnknown, num_channels);
+  SineWaveGenerator wave_generator(1000.0, audio_level);
+  wave_generator.GenerateNextFrame(audio_frame.get());
+  return audio_frame;
+}
+
+}  // namespace
+
+TEST(AudioSendStreamTest, ConfigToString) {
+  AudioSendStream::Config config(/*send_transport=*/nullptr);
+  config.rtp.ssrc = kSsrc;
+  config.rtp.c_name = kCName;
+  config.min_bitrate_bps = 12000;
+  config.max_bitrate_bps = 34000;
+  config.has_dscp = true;
+  config.send_codec_spec =
+      AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
+  config.send_codec_spec->nack_enabled = true;
+  config.send_codec_spec->transport_cc_enabled = false;
+  config.send_codec_spec->cng_payload_type = 42;
+  config.send_codec_spec->red_payload_type = 43;
+  config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
+  config.rtp.extmap_allow_mixed = true;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+  config.rtcp_report_interval_ms = 2500;
+  EXPECT_EQ(
+      "{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
+      "c_name: foo_name}, rtcp_report_interval_ms: 2500, "
+      "send_transport: null, "
+      "min_bitrate_bps: 12000, max_bitrate_bps: 34000, has "
+      "audio_network_adaptor_config: false, has_dscp: true, "
+      "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
+      "enable_non_sender_rtt: false, cng_payload_type: 42, "
+      "red_payload_type: 43, payload_type: 103, "
+      "format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
+      "parameters: {}}}}",
+      config.ToString());
+}
+
+TEST(AudioSendStreamTest, ConstructDestruct) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, SendTelephoneEvent) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForSendTelephoneEvent();
+    EXPECT_TRUE(send_stream->SendTelephoneEvent(
+        kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency,
+        kTelephoneEventCode, kTelephoneEventDuration));
+  }
+}
+
+TEST(AudioSendStreamTest, SetMuted) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(*helper.channel_send(), SetInputMute(true));
+    send_stream->SetMuted(true);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, GetStats) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForGetStats(use_null_audio_processing);
+    AudioSendStream::Stats stats = send_stream->GetStats(true);
+    EXPECT_EQ(kSsrc, stats.local_ssrc);
+    EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
+    EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
+              stats.header_and_padding_bytes_sent);
+    EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
+    EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
+    EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
+    EXPECT_EQ(kIsacFormat.name, stats.codec_name);
+    EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
+                                   (kIsacFormat.clockrate_hz / 1000)),
+              stats.jitter_ms);
+    EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
+    EXPECT_EQ(0, stats.audio_level);
+    EXPECT_EQ(0, stats.total_input_energy);
+    EXPECT_EQ(0, stats.total_input_duration);
+
+    if (!use_null_audio_processing) {
+      EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms);
+      EXPECT_EQ(kEchoDelayStdDev,
+                stats.apm_statistics.delay_standard_deviation_ms);
+      EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss);
+      EXPECT_EQ(kEchoReturnLossEnhancement,
+                stats.apm_statistics.echo_return_loss_enhancement);
+      EXPECT_EQ(kDivergentFilterFraction,
+                stats.apm_statistics.divergent_filter_fraction);
+      EXPECT_EQ(kResidualEchoLikelihood,
+                stats.apm_statistics.residual_echo_likelihood);
+      EXPECT_EQ(kResidualEchoLikelihoodMax,
+                stats.apm_statistics.residual_echo_likelihood_recent_max);
+    }
+  }
+}
+
+TEST(AudioSendStreamTest, GetStatsAudioLevel) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForGetStats(use_null_audio_processing);
+    EXPECT_CALL(*helper.channel_send(), ProcessAndEncodeAudio)
+        .Times(AnyNumber());
+
+    constexpr int kSampleRateHz = 48000;
+    constexpr size_t kNumChannels = 1;
+
+    constexpr int16_t kSilentAudioLevel = 0;
+    constexpr int16_t kMaxAudioLevel = 32767;  // Audio level is [0,32767].
+    constexpr int kAudioFrameDurationMs = 10;
+
+    // Process 10 audio frames (100 ms) of silence. After this, on the next
+    // (11-th) frame, the audio level will be updated with the maximum audio
+    // level of the first 11 frames. See AudioLevel.
+    for (size_t i = 0; i < 10; ++i) {
+      send_stream->SendAudioData(
+          CreateAudioFrame1kHzSineWave(kSilentAudioLevel, kAudioFrameDurationMs,
+                                       kSampleRateHz, kNumChannels));
+    }
+    AudioSendStream::Stats stats = send_stream->GetStats();
+    EXPECT_EQ(kSilentAudioLevel, stats.audio_level);
+    EXPECT_NEAR(0.0f, stats.total_input_energy, kTolerance);
+    EXPECT_NEAR(0.1f, stats.total_input_duration,
+                kTolerance);  // 100 ms = 0.1 s
+
+    // Process 10 audio frames (100 ms) of maximum audio level.
+    // Note that AudioLevel updates the audio level every 11th frame, processing
+    // 10 frames above was needed to see a non-zero audio level here.
+    for (size_t i = 0; i < 10; ++i) {
+      send_stream->SendAudioData(CreateAudioFrame1kHzSineWave(
+          kMaxAudioLevel, kAudioFrameDurationMs, kSampleRateHz, kNumChannels));
+    }
+    stats = send_stream->GetStats();
+    EXPECT_EQ(kMaxAudioLevel, stats.audio_level);
+    // Energy increases by energy*duration, where energy is audio level in
+    // [0,1].
+    EXPECT_NEAR(0.1f, stats.total_input_energy, kTolerance);  // 0.1 s of max
+    EXPECT_NEAR(0.2f, stats.total_input_duration,
+                kTolerance);  // 200 ms = 0.2 s
+  }
+}
+
+TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
+    const std::string kAnaConfigString = "abcde";
+    const std::string kAnaReconfigString = "12345";
+
+    helper.config().audio_network_adaptor_config = kAnaConfigString;
+
+    EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
+        .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
+                             int payload_type, const SdpAudioFormat& format,
+                             absl::optional<AudioCodecPairId> codec_pair_id,
+                             std::unique_ptr<AudioEncoder>* return_value) {
+          auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
+          EXPECT_CALL(*mock_encoder,
+                      EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
+              .WillOnce(Return(true));
+          EXPECT_CALL(*mock_encoder,
+                      EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _))
+              .WillOnce(Return(true));
+          *return_value = std::move(mock_encoder);
+        }));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    auto stream_config = helper.config();
+    stream_config.audio_network_adaptor_config = kAnaReconfigString;
+
+    send_stream->Reconfigure(stream_config, nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioNetworkAdaptorReceivesOverhead) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
+    const std::string kAnaConfigString = "abcde";
+
+    EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
+        .WillOnce(Invoke(
+            [&kAnaConfigString](int payload_type, const SdpAudioFormat& format,
+                                absl::optional<AudioCodecPairId> codec_pair_id,
+                                std::unique_ptr<AudioEncoder>* return_value) {
+              auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
+              InSequence s;
+              EXPECT_CALL(
+                  *mock_encoder,
+                  OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>())))
+                  .Times(2);
+              EXPECT_CALL(*mock_encoder,
+                          EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
+                  .WillOnce(Return(true));
+              // Note: Overhead is received AFTER ANA has been enabled.
+              EXPECT_CALL(
+                  *mock_encoder,
+                  OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>())))
+                  .WillOnce(Return());
+              *return_value = std::move(mock_encoder);
+            }));
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    auto stream_config = helper.config();
+    stream_config.audio_network_adaptor_config = kAnaConfigString;
+
+    send_stream->Reconfigure(stream_config, nullptr);
+  }
+}
+
+// VAD is applied when codec is mono and the CNG frequency matches the codec
+// clock rate.
+TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, false, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+    helper.config().send_codec_spec->cng_payload_type = 105;
+    std::unique_ptr<AudioEncoder> stolen_encoder;
+    EXPECT_CALL(*helper.channel_send(), SetEncoder)
+        .WillOnce([&stolen_encoder](int payload_type,
+                                    std::unique_ptr<AudioEncoder> encoder) {
+          stolen_encoder = std::move(encoder);
+          return true;
+        });
+    EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    // We cannot truly determine if the encoder created is an AudioEncoderCng.
+    // It is the only reasonable implementation that will return something from
+    // ReclaimContainedEncoders, though.
+    ASSERT_TRUE(stolen_encoder);
+    EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty());
+  }
+}
+
+TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(
+            Field(&BitrateAllocationUpdate::target_bitrate,
+                  Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
+    update.packet_loss_ratio = 0;
+    update.round_trip_time = TimeDelta::Millis(50);
+    update.bwe_period = TimeDelta::Millis(6000);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(
+            &BitrateAllocationUpdate::target_bitrate,
+            Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
+                                  Eq(DataRate::KilobitsPerSec(6)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(1);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
+                                  Eq(DataRate::KilobitsPerSec(64)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(128);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverhead) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(helper.field_trials,
+                                      "WebRTC-Audio-LegacyOverhead/Disabled/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = bitrate;
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials,
+        "WebRTC-Audio-LegacyOverhead/Disabled/"
+        "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate = DataRate::KilobitsPerSec(6) + kMinOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(1);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials,
+        "WebRTC-Audio-LegacyOverhead/Disabled/"
+        "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate = DataRate::KilobitsPerSec(64) + kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(128);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
+                                          Eq(TimeDelta::Millis(5000)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
+    update.packet_loss_ratio = 0;
+    update.round_trip_time = TimeDelta::Millis(50);
+    update.bwe_period = TimeDelta::Millis(5000);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+  }
+}
+
+// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
+TEST(AudioSendStreamTest, DontRecreateEncoder) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, false, use_null_audio_processing);
+    // WillOnce is (currently) the default used by ConfigHelper if asked to set
+    // an expectation for SetEncoder. Since this behavior is essential for this
+    // test to be correct, it's instead set-up manually here. Otherwise a simple
+    // change to ConfigHelper (say to WillRepeatedly) would silently make this
+    // test useless.
+    EXPECT_CALL(*helper.channel_send(), SetEncoder).WillOnce(Return());
+
+    EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+    helper.config().send_codec_spec->cng_payload_type = 105;
+    auto send_stream = helper.CreateAudioSendStream();
+    send_stream->Reconfigure(helper.config(), nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+    ConfigHelper::AddBweToConfig(&new_config);
+
+    EXPECT_CALL(*helper.rtp_rtcp(),
+                RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
+                                           kTransportSequenceNumberId))
+        .Times(1);
+    {
+      ::testing::InSequence seq;
+      EXPECT_CALL(*helper.channel_send(), ResetSenderCongestionControlObjects())
+          .Times(1);
+      EXPECT_CALL(*helper.channel_send(),
+                  RegisterSenderCongestionControlObjects(helper.transport(),
+                                                         Ne(nullptr)))
+          .Times(1);
+    }
+
+    send_stream->Reconfigure(new_config, nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, OnTransportOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    // CallEncoder will be called on overhead change.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    EXPECT_EQ(transport_overhead_per_packet_bytes,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+TEST(AudioSendStreamTest, DoesntCallEncoderWhenOverheadUnchanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    // CallEncoder will be called on overhead change.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    // Set the same overhead again, CallEncoder should not be called again.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder).Times(0);
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    // New overhead, call CallEncoder again
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes + 1);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    const size_t audio_overhead_per_packet_bytes = 555;
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes));
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+
+    EXPECT_EQ(audio_overhead_per_packet_bytes,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes + 20));
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+
+    EXPECT_EQ(audio_overhead_per_packet_bytes + 20,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+TEST(AudioSendStreamTest, OnAudioAndTransportOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    const size_t audio_overhead_per_packet_bytes = 555;
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes));
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    helper.worker()->RunSynchronous(
+        [&] { send_stream->OnBitrateUpdated(update); });
+
+    EXPECT_EQ(
+        transport_overhead_per_packet_bytes + audio_overhead_per_packet_bytes,
+        send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+// Validates that reconfiguring the AudioSendStream with a Frame encryptor
+// correctly reconfigures on the object without crashing.
+TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0(
+        rtc::make_ref_counted<MockFrameEncryptor>());
+    new_config.frame_encryptor = mock_frame_encryptor_0;
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
+        .Times(1);
+    send_stream->Reconfigure(new_config, nullptr);
+
+    // Not updating the frame encryptor shouldn't force it to reconfigure.
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0);
+    send_stream->Reconfigure(new_config, nullptr);
+
+    // Updating frame encryptor to a new object should force a call to the
+    // proxy.
+    rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1(
+        rtc::make_ref_counted<MockFrameEncryptor>());
+    new_config.frame_encryptor = mock_frame_encryptor_1;
+    new_config.crypto_options.sframe.require_frame_encryption = true;
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
+        .Times(1);
+    send_stream->Reconfigure(new_config, nullptr);
+  }
+}
+}  // namespace test
+}  // namespace webrtc