Update modules/ to not use implicit conversion from scoped_refptr<T> to T*.

Bug: webrtc:13464
Change-Id: I3906e91906edbf80d558e5c367d6b9429497c021
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259762
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36606}

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -320,7 +320,7 @@ TEST_F(NetEqImplTest, InsertPacket) {
         *dec = std::move(mock_decoder);
       }));
   DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                    absl::nullopt, mock_decoder_factory);
+                                    absl::nullopt, mock_decoder_factory.get());
 
   // Expectations for decoder database.
   EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
@@ -1633,7 +1633,7 @@ TEST_F(NetEqImplTest, NoCrashWith1000Channels) {
         decoder = dec->get();
       }));
   DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                    absl::nullopt, mock_decoder_factory);
+                                    absl::nullopt, mock_decoder_factory.get());
   // Expectations for decoder database.
   EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
       .WillRepeatedly(Return(&info));

modules/audio_coding/neteq/packet_buffer_unittest.cc

@@ -316,7 +316,7 @@ TEST(PacketBuffer, InsertPacketList) {
   MockDecoderDatabase decoder_database;
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(&info));
 
@@ -366,11 +366,11 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) {
   MockDecoderDatabase decoder_database;
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info0(SdpAudioFormat("pcmu", 8000, 1),
-                                           absl::nullopt, factory);
+                                           absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(&info0));
   const DecoderDatabase::DecoderInfo info1(SdpAudioFormat("pcma", 8000, 1),
-                                           absl::nullopt, factory);
+                                           absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(1))
       .WillRepeatedly(Return(&info1));
 
@@ -562,7 +562,7 @@ TEST(PacketBuffer, Reordering) {
   MockDecoderDatabase decoder_database;
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(&info));
   absl::optional<uint8_t> current_pt;
@@ -609,11 +609,11 @@ TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
   MockDecoderDatabase decoder_database;
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 8000, 1),
-                                              absl::nullopt, factory);
+                                              absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt))
       .WillRepeatedly(Return(&info_cng));
   const DecoderDatabase::DecoderInfo info_speech(
-      SdpAudioFormat("l16", 16000, 1), absl::nullopt, factory);
+      SdpAudioFormat("l16", 16000, 1), absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt))
       .WillRepeatedly(Return(&info_speech));
 
@@ -736,7 +736,7 @@ TEST(PacketBuffer, Failures) {
   list.push_back(gen.NextPacket(payload_len, nullptr));  // Valid packet.
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
       .WillRepeatedly(Return(&info));
   absl::optional<uint8_t> current_pt;

modules/audio_coding/neteq/timestamp_scaler_unittest.cc

@@ -27,7 +27,7 @@ TEST(TimestampScaler, TestNoScaling) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use PCMu, because it doesn't use scaled timestamps.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -49,7 +49,7 @@ TEST(TimestampScaler, TestNoScalingLargeStep) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use PCMu, because it doesn't use scaled timestamps.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 0;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -76,7 +76,7 @@ TEST(TimestampScaler, TestG722) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -102,7 +102,7 @@ TEST(TimestampScaler, TestG722LargeStep) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -132,9 +132,9 @@ TEST(TimestampScaler, TestG722WithCng) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info_g722(SdpAudioFormat("g722", 8000, 1),
-                                               absl::nullopt, factory);
+                                               absl::nullopt, factory.get());
   const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 16000, 1),
-                                              absl::nullopt, factory);
+                                              absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadTypeG722 = 17;
   static const uint8_t kRtpPayloadTypeCng = 13;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
@@ -176,7 +176,7 @@ TEST(TimestampScaler, TestG722Packet) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -206,7 +206,7 @@ TEST(TimestampScaler, TestG722PacketList) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -240,7 +240,7 @@ TEST(TimestampScaler, TestG722Reset) {
   auto factory = CreateBuiltinAudioDecoderFactory();
   // Use G722, which has a factor 2 scaling.
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));
@@ -281,7 +281,7 @@ TEST(TimestampScaler, TestOpusLargeStep) {
   MockDecoderDatabase db;
   auto factory = CreateBuiltinAudioDecoderFactory();
   const DecoderDatabase::DecoderInfo info(SdpAudioFormat("opus", 48000, 2),
-                                          absl::nullopt, factory);
+                                          absl::nullopt, factory.get());
   static const uint8_t kRtpPayloadType = 17;
   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
       .WillRepeatedly(Return(&info));

modules/audio_device/win/audio_device_core_win.cc

@@ -23,7 +23,11 @@
 
 #ifdef WEBRTC_WINDOWS_CORE_AUDIO_BUILD
 
+// clang-format off
+// To get Windows includes in the right order, this must come before the Windows
+// includes below.
 #include "modules/audio_device/win/audio_device_core_win.h"
+// clang-format on
 
 #include <string.h>
 
@@ -40,6 +44,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/platform_thread.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/string_utils.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/sleep.h"
@@ -2090,7 +2095,8 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO() {
         << "AudioDeviceBuffer must be attached before streaming can start";
   }
 
-  _mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
+  _mediaBuffer = rtc::make_ref_counted<MediaBufferImpl>(_recBlockSize *
+                                                        _recAudioFrameSize);
 
   // Optional, but if called, must be after media types are set.
   hr = _dmo->AllocateStreamingResources();
@@ -2996,7 +3002,7 @@ DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO() {
       DWORD dwStatus = 0;
       {
         DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
-        dmoBuffer.pBuffer = _mediaBuffer;
+        dmoBuffer.pBuffer = _mediaBuffer.get();
         dmoBuffer.pBuffer->AddRef();
 
         // Poll the DMO for AEC processed capture data. The DMO will
@@ -3393,32 +3399,34 @@ int AudioDeviceWindowsCore::SetDMOProperties() {
 
   // Set the AEC system mode.
   // SINGLE_CHANNEL_AEC - AEC processing only.
-  if (SetVtI4Property(ps, MFPKEY_WMAAECMA_SYSTEM_MODE, SINGLE_CHANNEL_AEC)) {
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_SYSTEM_MODE,
+                      SINGLE_CHANNEL_AEC)) {
     return -1;
   }
 
   // Set the AEC source mode.
   // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
-  if (SetBoolProperty(ps, MFPKEY_WMAAECMA_DMO_SOURCE_MODE, VARIANT_TRUE) ==
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
-      -1) {
+                      VARIANT_TRUE) == -1) {
     return -1;
   }
 
   // Enable the feature mode.
   // This lets us override all the default processing settings below.
-  if (SetBoolProperty(ps, MFPKEY_WMAAECMA_FEATURE_MODE, VARIANT_TRUE) == -1) {
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_FEATURE_MODE, VARIANT_TRUE) ==
+      -1) {
     return -1;
   }
 
   // Disable analog AGC (default enabled).
-  if (SetBoolProperty(ps, MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER, VARIANT_FALSE) ==
+  if (SetBoolProperty(ps.get(), MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
-      -1) {
+                      VARIANT_FALSE) == -1) {
     return -1;
   }
 
   // Disable noise suppression (default enabled).
   // 0 - Disabled, 1 - Enabled
-  if (SetVtI4Property(ps, MFPKEY_WMAAECMA_FEATR_NS, 0) == -1) {
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_FEATR_NS, 0) == -1) {
     return -1;
   }
 
@@ -3463,7 +3471,8 @@ int AudioDeviceWindowsCore::SetDMOProperties() {
                    static_cast<uint32_t>(0x0000ffff & inDevIndex);
   RTC_LOG(LS_VERBOSE) << "Capture device index: " << inDevIndex
                       << ", render device index: " << outDevIndex;
-  if (SetVtI4Property(ps, MFPKEY_WMAAECMA_DEVICE_INDEXES, devIndex) == -1) {
+  if (SetVtI4Property(ps.get(), MFPKEY_WMAAECMA_DEVICE_INDEXES, devIndex) ==
+      -1) {
     return -1;
   }
 
@@ -3766,7 +3775,7 @@ int32_t AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
       SAFE_RELEASE(ptrDevice);
     }
 
-    if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1) {
+    if (_GetDeviceID(device.get(), szDeviceID, kDeviceIDLength) == -1) {
       return -1;
     }
 

modules/rtp_rtcp/source/forward_error_correction.cc

@@ -651,10 +651,10 @@ bool ForwardErrorCorrection::RecoverPacket(const ReceivedFecPacket& fec_packet,
       // This is the packet we're recovering.
       recovered_packet->seq_num = protected_packet->seq_num;
     } else {
-      XorHeaders(*protected_packet->pkt, recovered_packet->pkt);
+      XorHeaders(*protected_packet->pkt, recovered_packet->pkt.get());
       XorPayloads(*protected_packet->pkt,
                   protected_packet->pkt->data.size() - kRtpHeaderSize,
-                  kRtpHeaderSize, recovered_packet->pkt);
+                  kRtpHeaderSize, recovered_packet->pkt.get());
     }
   }
   if (!FinishPacketRecovery(fec_packet, recovered_packet)) {

modules/rtp_rtcp/source/rtp_sender_video_unittest.cc

@@ -787,7 +787,7 @@ TEST_P(RtpSenderVideoTest,
   config.clock = &fake_clock_;
   config.rtp_sender = rtp_module_->RtpSender();
   config.field_trials = &field_trials_;
-  config.frame_encryptor = encryptor;
+  config.frame_encryptor = encryptor.get();
   RTPSenderVideo rtp_sender_video(config);
 
   FrameDependencyStructure video_structure;

modules/rtp_rtcp/source/ulpfec_receiver_impl.cc

@@ -177,7 +177,7 @@ int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
   for (const auto& received_packet : received_packets) {
     // Send received media packet to VCM.
     if (!received_packet->is_fec) {
-      ForwardErrorCorrection::Packet* packet = received_packet->pkt;
+      ForwardErrorCorrection::Packet* packet = received_packet->pkt.get();
       recovered_packet_callback_->OnRecoveredPacket(packet->data.data(),
                                                     packet->data.size());
       // Create a packet with the buffer to modify it.
@@ -211,7 +211,7 @@ int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
       // Already sent to the VCM and the jitter buffer.
       continue;
     }
-    ForwardErrorCorrection::Packet* packet = recovered_packet->pkt;
+    ForwardErrorCorrection::Packet* packet = recovered_packet->pkt.get();
     ++packet_counter_.num_recovered_packets;
     // Set this flag first; in case the recovered packet carries a RED
     // header, OnRecoveredPacket will recurse back here.

modules/video_capture/windows/sink_filter_ds.cc

@@ -707,7 +707,7 @@ CaptureInputPin::GetAllocator(IMemAllocator** allocator) {
       return hr;
     allocator_.swap(allocator);
   }
-  *allocator = allocator_;
+  *allocator = allocator_.get();
   allocator_->AddRef();
   return S_OK;
 }

modules/video_capture/windows/video_capture_ds.cc

@@ -35,7 +35,7 @@ VideoCaptureDS::~VideoCaptureDS() {
   }
   if (_graphBuilder) {
     if (sink_filter_)
-      _graphBuilder->RemoveFilter(sink_filter_);
+      _graphBuilder->RemoveFilter(sink_filter_.get());
     if (_captureFilter)
       _graphBuilder->RemoveFilter(_captureFilter);
     if (_dvFilter)
@@ -101,13 +101,13 @@ int32_t VideoCaptureDS::Init(const char* deviceUniqueIdUTF8) {
   // Create the sink filte used for receiving Captured frames.
   sink_filter_ = new ComRefCount<CaptureSinkFilter>(this);
 
-  hr = _graphBuilder->AddFilter(sink_filter_, SINK_FILTER_NAME);
+  hr = _graphBuilder->AddFilter(sink_filter_.get(), SINK_FILTER_NAME);
   if (FAILED(hr)) {
     RTC_LOG(LS_INFO) << "Failed to add the send filter to the graph.";
     return -1;
   }
 
-  _inputSendPin = GetInputPin(sink_filter_);
+  _inputSendPin = GetInputPin(sink_filter_.get());
   if (!_inputSendPin) {
     RTC_LOG(LS_INFO) << "Failed to get input send pin";
     return -1;

modules/video_coding/codecs/test/videoprocessor.cc

@@ -507,7 +507,7 @@ void VideoProcessor::WriteDecodedFrame(const I420BufferInterface& decoded_frame,
     scaled_buffer = I420Buffer::Create(input_video_width, input_video_height);
     scaled_buffer->ScaleFrom(decoded_frame);
 
-    scaled_frame = scaled_buffer;
+    scaled_frame = scaled_buffer.get();
   }
 
   // Ensure there is no padding.

modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc

@@ -1367,7 +1367,7 @@ LibvpxVp8Encoder::PrepareBuffers(rtc::scoped_refptr<VideoFrameBuffer> buffer) {
   // Prepare `raw_images_` from `mapped_buffer` and, if simulcast, scaled
   // versions of `buffer`.
   std::vector<rtc::scoped_refptr<VideoFrameBuffer>> prepared_buffers;
-  SetRawImagePlanes(&raw_images_[0], mapped_buffer);
+  SetRawImagePlanes(&raw_images_[0], mapped_buffer.get());
   prepared_buffers.push_back(mapped_buffer);
   for (size_t i = 1; i < encoders_.size(); ++i) {
     // Native buffers should implement optimized scaling and is the preferred
@@ -1410,7 +1410,7 @@ LibvpxVp8Encoder::PrepareBuffers(rtc::scoped_refptr<VideoFrameBuffer> buffer) {
           << VideoFrameBufferTypeToString(mapped_buffer->type());
       return {};
     }
-    SetRawImagePlanes(&raw_images_[i], scaled_buffer);
+    SetRawImagePlanes(&raw_images_[i], scaled_buffer.get());
     prepared_buffers.push_back(scaled_buffer);
   }
   return prepared_buffers;