Fix ClangTidy issues in video/

These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.

Bug: webrtc:10410
Change-Id: Iedb3be944828a1caba55bbbd4dc0b56c55bbb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127624
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27123}

video/call_stats_unittest.cc

@@ -21,7 +21,6 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 
-using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::InvokeWithoutArgs;
 using ::testing::Return;

video/encoder_key_frame_callback_unittest.cc

@@ -16,8 +16,6 @@
 #include "test/gtest.h"
 #include "video/test/mock_video_stream_encoder.h"
 
-using ::testing::NiceMock;
-
 namespace webrtc {
 
 class VieKeyRequestTest : public ::testing::Test {

video/frame_encode_timer.cc

@@ -203,7 +203,7 @@ absl::optional<int64_t> FrameEncodeTimer::ExtractEncodeStartTime(
           EncodedImageCallback::DropReason::kDroppedByEncoder);
       encode_start_list->pop_front();
     }
-    if (encode_start_list->size() > 0 &&
+    if (!encode_start_list->empty() &&
         encode_start_list->front().rtp_timestamp ==
             encoded_image->Timestamp()) {
       result.emplace(encode_start_list->front().encode_start_time_ms);

video/video_quality_test.cc

@@ -1012,7 +1012,7 @@ std::unique_ptr<test::FrameGenerator> VideoQualityTest::CreateFrameGenerator(
             params_.video[video_idx].fps);
   } else {
     std::vector<std::string> slides = params_.screenshare[video_idx].slides;
-    if (slides.size() == 0) {
+    if (slides.empty()) {
       slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
       slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
       slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
@@ -1206,8 +1206,8 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
   task_queue_.SendTask([this, &send_call_config, &recv_call_config,
                         &send_transport, &recv_transport]() {
     if (params_.audio.enabled)
-      InitializeAudioDevice(
-          &send_call_config, &recv_call_config, params_.audio.use_real_adm);
+      InitializeAudioDevice(&send_call_config, &recv_call_config,
+                            params_.audio.use_real_adm);
 
     CreateCalls(send_call_config, recv_call_config);
     send_transport = CreateSendTransport();
@@ -1298,21 +1298,21 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
 
 rtc::scoped_refptr<AudioDeviceModule> VideoQualityTest::CreateAudioDevice() {
 #ifdef WEBRTC_WIN
-    RTC_LOG(INFO) << "Using latest version of ADM on Windows";
-    // We must initialize the COM library on a thread before we calling any of
-    // the library functions. All COM functions in the ADM will return
-    // CO_E_NOTINITIALIZED otherwise. The legacy ADM for Windows used internal
-    // COM initialization but the new ADM requires COM to be initialized
-    // externally.
-    com_initializer_ = absl::make_unique<webrtc_win::ScopedCOMInitializer>(
-        webrtc_win::ScopedCOMInitializer::kMTA);
-    RTC_CHECK(com_initializer_->Succeeded());
-    RTC_CHECK(webrtc_win::core_audio_utility::IsSupported());
-    RTC_CHECK(webrtc_win::core_audio_utility::IsMMCSSSupported());
-    return CreateWindowsCoreAudioAudioDeviceModule();
+  RTC_LOG(INFO) << "Using latest version of ADM on Windows";
+  // We must initialize the COM library on a thread before we calling any of
+  // the library functions. All COM functions in the ADM will return
+  // CO_E_NOTINITIALIZED otherwise. The legacy ADM for Windows used internal
+  // COM initialization but the new ADM requires COM to be initialized
+  // externally.
+  com_initializer_ = absl::make_unique<webrtc_win::ScopedCOMInitializer>(
+      webrtc_win::ScopedCOMInitializer::kMTA);
+  RTC_CHECK(com_initializer_->Succeeded());
+  RTC_CHECK(webrtc_win::core_audio_utility::IsSupported());
+  RTC_CHECK(webrtc_win::core_audio_utility::IsMMCSSSupported());
+  return CreateWindowsCoreAudioAudioDeviceModule();
 #else
-    // Use legacy factory method on all platforms except Windows.
-    return AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio);
+  // Use legacy factory method on all platforms except Windows.
+  return AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio);
 #endif
 }
 
@@ -1348,7 +1348,7 @@ void VideoQualityTest::InitializeAudioDevice(Call::Config* send_call_config,
   }
   // Always initialize the ADM before injecting a valid audio transport.
   RTC_CHECK(audio_device->RegisterAudioCallback(
-            send_call_config->audio_state->audio_transport()) == 0);
+                send_call_config->audio_state->audio_transport()) == 0);
 }
 
 void VideoQualityTest::SetupAudio(Transport* transport) {
@@ -1428,8 +1428,8 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
     Call::Config recv_call_config(recv_event_log_.get());
 
     if (params_.audio.enabled)
-      InitializeAudioDevice(
-          &send_call_config, &recv_call_config, params_.audio.use_real_adm);
+      InitializeAudioDevice(&send_call_config, &recv_call_config,
+                            params_.audio.use_real_adm);
 
     CreateCalls(send_call_config, recv_call_config);
 

video/video_send_stream.cc

@@ -34,7 +34,7 @@ size_t CalculateMaxHeaderSize(const RtpConfig& config) {
   size_t header_size = kRtpHeaderSize;
   size_t extensions_size = 0;
   size_t fec_extensions_size = 0;
-  if (config.extensions.size() > 0) {
+  if (!config.extensions.empty()) {
     RtpHeaderExtensionMap extensions_map(config.extensions);
     extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(),
                                              extensions_map);

video/video_send_stream_impl_unittest.cc

@@ -30,12 +30,10 @@
 namespace webrtc {
 namespace internal {
 namespace {
-using testing::NiceMock;
-using testing::StrictMock;
-using testing::ReturnRef;
-using testing::Return;
-using testing::Invoke;
 using testing::_;
+using testing::Invoke;
+using testing::NiceMock;
+using testing::Return;
 
 constexpr int64_t kDefaultInitialBitrateBps = 333000;
 const double kDefaultBitratePriority = 0.5;

video/video_stream_encoder.cc

@@ -817,7 +817,7 @@ void VideoStreamEncoder::ConfigureQualityScaler(
       scaling_settings.thresholds;
 
   if (quality_scaling_allowed) {
-    if (quality_scaler_.get() == nullptr) {
+    if (quality_scaler_ == nullptr) {
       // Quality scaler has not already been configured.
 
       // Use experimental thresholds if available.

video/video_stream_encoder_unittest.cc

@@ -42,7 +42,6 @@ namespace webrtc {
 
 using ScaleReason = AdaptationObserverInterface::AdaptReason;
 using ::testing::_;
-using ::testing::Return;
 
 namespace {
 const int kMinPixelsPerFrame = 320 * 180;