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Update audio/, media/, and video/ to not use implicit conversion

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from scoped_refptr<T> to T*.

Bug: webrtc:13464
Change-Id: Ia14885f359fea2bdf08a41b3ded82532a9585d34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259503
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36599}
This commit is contained in:
Niels Möller 2022-04-20 16:46:26 +02:00 committed by WebRTC LUCI CQ
parent 3a8829a183
commit ba2de58a22
10 changed files with 20 additions and 20 deletions
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2
audio/audio_send_stream.cc
View File

@ -122,7 +122,7 @@ AudioSendStream::AudioSendStream(
config.send_transport,
rtcp_rtt_stats,
event_log,
config.frame_encryptor,
config.frame_encryptor.get(),
config.crypto_options,
config.rtp.extmap_allow_mixed,
config.rtcp_report_interval_ms,

2
audio/audio_state.cc
View File

@ -28,7 +28,7 @@ namespace internal {
AudioState::AudioState(const AudioState::Config& config)
: config_(config),
audio_transport_(config_.audio_mixer,
audio_transport_(config_.audio_mixer.get(),
config_.audio_processing.get(),
config_.async_audio_processing_factory.get()) {
process_thread_checker_.Detach();

2
media/base/video_broadcaster_unittest.cc
View File

@ -66,7 +66,7 @@ TEST(VideoBroadcasterTest, OnFrame) {
rtc::scoped_refptr<webrtc::I420Buffer> buffer(
webrtc::I420Buffer::Create(kWidth, kHeight));
// Initialize, to avoid warnings on use of initialized values.
webrtc::I420Buffer::SetBlack(buffer);
webrtc::I420Buffer::SetBlack(buffer.get());
webrtc::VideoFrame frame = webrtc::VideoFrame::Builder()
.set_video_frame_buffer(buffer)

4
media/engine/fake_webrtc_call.cc
View File

@ -259,7 +259,7 @@ void FakeVideoSendStream::ReconfigureVideoEncoder(
}
video_streams_ =
config.video_stream_factory->CreateEncoderStreams(width, height, config);
if (config.encoder_specific_settings != NULL) {
if (config.encoder_specific_settings != nullptr) {
const unsigned char num_temporal_layers = static_cast<unsigned char>(
video_streams_.back().num_temporal_layers.value_or(1));
if (config_.rtp.payload_name == "VP8") {
@ -286,7 +286,7 @@ void FakeVideoSendStream::ReconfigureVideoEncoder(
<< config_.rtp.payload_name;
}
}
codec_settings_set_ = config.encoder_specific_settings != NULL;
codec_settings_set_ = config.encoder_specific_settings != nullptr;
encoder_config_ = std::move(config);
++num_encoder_reconfigurations_;
}

2
media/engine/null_webrtc_video_engine_unittest.cc
View File

@ -34,7 +34,7 @@ TEST(NullWebRtcVideoEngineTest, CheckInterface) {
webrtc::test::MockAudioDeviceModule::CreateNice();
webrtc::FieldTrialBasedConfig trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
webrtc::AudioProcessingBuilder().Create(), nullptr, trials);

2
media/engine/webrtc_media_engine.cc
View File

@ -35,7 +35,7 @@ std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
const webrtc::FieldTrialsView& trials =
dependencies.trials ? *dependencies.trials : *fallback_trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
dependencies.task_queue_factory, std::move(dependencies.adm),
dependencies.task_queue_factory, dependencies.adm.get(),
std::move(dependencies.audio_encoder_factory),
std::move(dependencies.audio_decoder_factory),
std::move(dependencies.audio_mixer),

2
media/engine/webrtc_video_engine_unittest.cc
View File

@ -174,7 +174,7 @@ rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
int height) {
rtc::scoped_refptr<webrtc::I420Buffer> buffer =
webrtc::I420Buffer::Create(width, height);
webrtc::I420Buffer::SetBlack(buffer);
webrtc::I420Buffer::SetBlack(buffer.get());
return buffer;
}

20
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -145,7 +145,7 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
webrtc::CreateDefaultTaskQueueFactory();
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
webrtc::test::MockAudioDeviceModule::CreateStrict();
AdmSetupExpectations(adm);
AdmSetupExpectations(adm.get());
rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm =
use_null_apm ? nullptr
: rtc::make_ref_counted<
@ -160,7 +160,7 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
{
webrtc::FieldTrialBasedConfig trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, trials);
@ -190,7 +190,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
StrictMock<webrtc::test::MockAudioProcessing>>()),
call_(&field_trials_) {
// AudioDeviceModule.
AdmSetupExpectations(adm_);
AdmSetupExpectations(adm_.get());
if (!use_null_apm_) {
// AudioProcessing.
@ -208,7 +208,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
engine_.reset(new cricket::WebRtcVoiceEngine(
task_queue_factory_.get(), adm_, encoder_factory, decoder_factory,
task_queue_factory_.get(), adm_.get(), encoder_factory, decoder_factory,
nullptr, apm_, nullptr, field_trials_));
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
@ -3611,7 +3611,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, field_trials);
@ -3641,7 +3641,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, field_trials);
@ -3676,7 +3676,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, field_trials);
@ -3729,7 +3729,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, field_trials);
@ -3778,7 +3778,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr,
field_trials);
@ -3831,7 +3831,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm, unused_encoder_factory,
task_queue_factory.get(), adm.get(), unused_encoder_factory,
mock_decoder_factory, nullptr, apm, nullptr, field_trials);
engine.Init();
auto codecs = engine.recv_codecs();

2
video/video_send_stream.cc
View File

@ -84,7 +84,7 @@ GetBitrateAllocationCallbackType(const VideoSendStream::Config& config,
RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig(
const VideoSendStream::Config* config) {
RtpSenderFrameEncryptionConfig frame_encryption_config;
frame_encryption_config.frame_encryptor = config->frame_encryptor;
frame_encryption_config.frame_encryptor = config->frame_encryptor.get();
frame_encryption_config.crypto_options = config->crypto_options;
return frame_encryption_config;
}

2
video/video_stream_encoder_unittest.cc
View File

@ -205,7 +205,7 @@ class FakeNV12NativeBuffer : public webrtc::VideoFrameBuffer {
}
return nullptr;
}
const NV12BufferInterface* GetNV12() const { return nv12_buffer_; }
const NV12BufferInterface* GetNV12() const { return nv12_buffer_.get(); }
private:
friend class rtc::RefCountedObject<FakeNV12NativeBuffer>;