Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

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

1 files changed, 366 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000000..4426a782d7
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,366 @@
+/*
+ *  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_state.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/test/mock_task_queue_base.h"
+#include "call/test/mock_audio_send_stream.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::Matcher;
+using ::testing::NiceMock;
+using ::testing::StrictMock;
+using ::testing::Values;
+
+constexpr int kSampleRate = 16000;
+constexpr int kNumberOfChannels = 1;
+
+struct FakeAsyncAudioProcessingHelper {
+  class FakeTaskQueue : public StrictMock<MockTaskQueueBase> {
+   public:
+    FakeTaskQueue() = default;
+
+    void Delete() override { delete this; }
+    void PostTask(absl::AnyInvocable<void() &&> task) override {
+      std::move(task)();
+    }
+  };
+
+  class FakeTaskQueueFactory : public TaskQueueFactory {
+   public:
+    FakeTaskQueueFactory() = default;
+    ~FakeTaskQueueFactory() override = default;
+    std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+        absl::string_view name,
+        Priority priority) const override {
+      return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
+          new FakeTaskQueue());
+    }
+  };
+
+  class MockAudioFrameProcessor : public AudioFrameProcessor {
+   public:
+    ~MockAudioFrameProcessor() override = default;
+
+    MOCK_METHOD(void, ProcessCalled, ());
+    MOCK_METHOD(void, SinkSet, ());
+    MOCK_METHOD(void, SinkCleared, ());
+
+    void Process(std::unique_ptr<AudioFrame> frame) override {
+      ProcessCalled();
+      sink_callback_(std::move(frame));
+    }
+
+    void SetSink(OnAudioFrameCallback sink_callback) override {
+      sink_callback_ = std::move(sink_callback);
+      if (sink_callback_ == nullptr)
+        SinkCleared();
+      else
+        SinkSet();
+    }
+
+   private:
+    OnAudioFrameCallback sink_callback_;
+  };
+
+  NiceMock<MockAudioFrameProcessor> audio_frame_processor_;
+  FakeTaskQueueFactory task_queue_factory_;
+
+  rtc::scoped_refptr<AsyncAudioProcessing::Factory> CreateFactory() {
+    return rtc::make_ref_counted<AsyncAudioProcessing::Factory>(
+        audio_frame_processor_, task_queue_factory_);
+  }
+};
+
+struct ConfigHelper {
+  struct Params {
+    bool use_null_audio_processing;
+    bool use_async_audio_processing;
+  };
+
+  explicit ConfigHelper(const Params& params)
+      : audio_mixer(AudioMixerImpl::Create()) {
+    audio_state_config.audio_mixer = audio_mixer;
+    audio_state_config.audio_processing =
+        params.use_null_audio_processing
+            ? nullptr
+            : rtc::make_ref_counted<testing::NiceMock<MockAudioProcessing>>();
+    audio_state_config.audio_device_module =
+        rtc::make_ref_counted<NiceMock<MockAudioDeviceModule>>();
+    if (params.use_async_audio_processing) {
+      audio_state_config.async_audio_processing_factory =
+          async_audio_processing_helper_.CreateFactory();
+    }
+  }
+  AudioState::Config& config() { return audio_state_config; }
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
+  NiceMock<FakeAsyncAudioProcessingHelper::MockAudioFrameProcessor>&
+  mock_audio_frame_processor() {
+    return async_audio_processing_helper_.audio_frame_processor_;
+  }
+
+ private:
+  AudioState::Config audio_state_config;
+  rtc::scoped_refptr<AudioMixer> audio_mixer;
+  FakeAsyncAudioProcessingHelper async_audio_processing_helper_;
+};
+
+class FakeAudioSource : public AudioMixer::Source {
+ public:
+  // TODO(aleloi): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  int Ssrc() const /*override*/ { return 0; }
+
+  int PreferredSampleRate() const /*override*/ { return kSampleRate; }
+
+  MOCK_METHOD(AudioFrameInfo,
+              GetAudioFrameWithInfo,
+              (int sample_rate_hz, AudioFrame*),
+              (override));
+};
+
+std::vector<int16_t> Create10msTestData(int sample_rate_hz,
+                                        size_t num_channels) {
+  const int samples_per_channel = sample_rate_hz / 100;
+  std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
+  // Fill the first channel with a 1kHz sine wave.
+  const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
+  float w = 0.f;
+  for (int i = 0; i < samples_per_channel; ++i) {
+    audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
+    w += inc;
+  }
+  return audio_data;
+}
+
+std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
+  const size_t num_channels = audio_frame->num_channels_;
+  const size_t samples_per_channel = audio_frame->samples_per_channel_;
+  std::vector<uint32_t> levels(num_channels, 0);
+  for (size_t i = 0; i < samples_per_channel; ++i) {
+    for (size_t j = 0; j < num_channels; ++j) {
+      levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
+    }
+  }
+  return levels;
+}
+}  // namespace
+
+class AudioStateTest : public ::testing::TestWithParam<ConfigHelper::Params> {};
+
+TEST_P(AudioStateTest, Create) {
+  ConfigHelper helper(GetParam());
+  auto audio_state = AudioState::Create(helper.config());
+  EXPECT_TRUE(audio_state.get());
+}
+
+TEST_P(AudioStateTest, ConstructDestruct) {
+  ConfigHelper helper(GetParam());
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+}
+
+TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  MockAudioSendStream stream;
+  audio_state->AddSendingStream(&stream, 8000, 2);
+
+  EXPECT_CALL(
+      stream,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(8000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(2u)))))
+      .WillOnce(
+          // Verify that channels are not swapped by default.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+            EXPECT_EQ(0u, levels[1]);
+          }));
+  MockAudioProcessing* ap =
+      GetParam().use_null_audio_processing
+          ? nullptr
+          : static_cast<MockAudioProcessing*>(audio_state->audio_processing());
+  if (ap) {
+    EXPECT_CALL(*ap, set_stream_delay_ms(0));
+    EXPECT_CALL(*ap, set_stream_key_pressed(false));
+    EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
+  }
+
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 2;
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream);
+}
+
+TEST_P(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  MockAudioSendStream stream_1;
+  MockAudioSendStream stream_2;
+  audio_state->AddSendingStream(&stream_1, 8001, 2);
+  audio_state->AddSendingStream(&stream_2, 32000, 1);
+
+  EXPECT_CALL(
+      stream_1,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
+      .WillOnce(
+          // Verify that there is output signal.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+          }));
+  EXPECT_CALL(
+      stream_2,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
+      .WillOnce(
+          // Verify that there is output signal.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+          }));
+  MockAudioProcessing* ap =
+      static_cast<MockAudioProcessing*>(audio_state->audio_processing());
+  if (ap) {
+    EXPECT_CALL(*ap, set_stream_delay_ms(5));
+    EXPECT_CALL(*ap, set_stream_key_pressed(true));
+    EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
+  }
+
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 1;
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 5, 0, 0, true, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream_1);
+  audio_state->RemoveSendingStream(&stream_2);
+}
+
+TEST_P(AudioStateTest, EnableChannelSwap) {
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 2;
+
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  audio_state->SetStereoChannelSwapping(true);
+
+  MockAudioSendStream stream;
+  audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);
+
+  EXPECT_CALL(stream, SendAudioDataForMock(_))
+      .WillOnce(
+          // Verify that channels are swapped.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_EQ(0u, levels[0]);
+            EXPECT_LT(0u, levels[1]);
+          }));
+
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream);
+}
+
+TEST_P(AudioStateTest,
+       QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
+  ConfigHelper helper(GetParam());
+  auto audio_state = AudioState::Create(helper.config());
+
+  FakeAudioSource fake_source;
+  helper.mixer()->AddSource(&fake_source);
+
+  EXPECT_CALL(fake_source, GetAudioFrameWithInfo(_, _))
+      .WillOnce(
+          ::testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
+            audio_frame->sample_rate_hz_ = sample_rate_hz;
+            audio_frame->samples_per_channel_ = sample_rate_hz / 100;
+            audio_frame->num_channels_ = kNumberOfChannels;
+            return AudioMixer::Source::AudioFrameInfo::kNormal;
+          }));
+
+  int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
+  size_t n_samples_out;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+  audio_state->audio_transport()->NeedMorePlayData(
+      kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
+      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
+}
+
+INSTANTIATE_TEST_SUITE_P(AudioStateTest,
+                         AudioStateTest,
+                         Values(ConfigHelper::Params({false, false}),
+                                ConfigHelper::Params({true, false}),
+                                ConfigHelper::Params({false, true}),
+                                ConfigHelper::Params({true, true})));
+
+}  // namespace test
+}  // namespace webrtc