zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Use absl::make_unique and absl::WrapUnique directly

Browse Source 
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
This commit is contained in:
Karl Wiberg
2018-07-05 11:40:33 +02:00
committed by Commit Bot
parent 431f14ef69
commit 918f50c5d1
322 changed files with 1190 additions and 1057 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
sdk/android/src/jni/audio_device/aaudio_player.cc
View File

@ -10,11 +10,11 @@
#include "sdk/android/src/jni/audio_device/aaudio_player.h"
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {
@ -124,7 +124,7 @@ void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
// Create a modified audio buffer class which allows us to ask for any number
// of samples (and not only multiple of 10ms) to match the optimal buffer
// size per callback used by AAudio.
fine_audio_buffer_ = rtc::MakeUnique<FineAudioBuffer>(audio_device_buffer_);
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
}
bool AAudioPlayer::SpeakerVolumeIsAvailable() {

4
sdk/android/src/jni/audio_device/aaudio_recorder.cc
View File

@ -10,11 +10,11 @@
#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/sleep.h"
@ -120,7 +120,7 @@ void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
// Create a modified audio buffer class which allows us to deliver any number
// of samples (and not only multiples of 10ms which WebRTC uses) to match the
// native AAudio buffer size.
fine_audio_buffer_ = rtc::MakeUnique<FineAudioBuffer>(audio_device_buffer_);
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
}
bool AAudioRecorder::IsAcousticEchoCancelerSupported() const {

2
sdk/android/src/jni/audio_device/audio_device_module.cc
View File

@ -86,7 +86,7 @@ class AndroidAudioDeviceModule : public AudioDeviceModule {
int32_t Init() override {
RTC_LOG(INFO) << __FUNCTION__;
RTC_DCHECK(thread_checker_.CalledOnValidThread());
audio_device_buffer_ = rtc::MakeUnique<AudioDeviceBuffer>();
audio_device_buffer_ = absl::make_unique<AudioDeviceBuffer>();
AttachAudioBuffer();
if (initialized_) {
return 0;

6
sdk/android/src/jni/audio_device/java_audio_device_module.cc
View File

@ -31,11 +31,11 @@ static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule(
GetAudioParameters(env, j_context, j_audio_manager, sample_rate,
j_use_stereo_input, j_use_stereo_output, &input_parameters,
&output_parameters);
auto audio_input = rtc::MakeUnique<AudioRecordJni>(
auto audio_input = absl::make_unique<AudioRecordJni>(
env, input_parameters, kHighLatencyModeDelayEstimateInMilliseconds,
j_webrtc_audio_record);
auto audio_output = rtc::MakeUnique<AudioTrackJni>(env, output_parameters,
j_webrtc_audio_track);
auto audio_output = absl::make_unique<AudioTrackJni>(env, output_parameters,
j_webrtc_audio_track);
return jlongFromPointer(CreateAudioDeviceModuleFromInputAndOutput(
AudioDeviceModule::kAndroidJavaAudio,
j_use_stereo_input, j_use_stereo_output,

4
sdk/android/src/jni/audio_device/opensles_player.cc
View File

@ -12,13 +12,13 @@
#include <android/log.h>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
@ -226,7 +226,7 @@ void OpenSLESPlayer::AllocateDataBuffers() {
ALOGD("native buffer size: %" PRIuS, buffer_size_in_samples);
ALOGD("native buffer size in ms: %.2f",
audio_parameters_.GetBufferSizeInMilliseconds());
fine_audio_buffer_ = rtc::MakeUnique<FineAudioBuffer>(audio_device_buffer_);
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
// Allocated memory for audio buffers.
for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);

4
sdk/android/src/jni/audio_device/opensles_recorder.cc
View File

@ -12,13 +12,13 @@
#include <android/log.h>
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "modules/audio_device/fine_audio_buffer.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "sdk/android/src/jni/audio_device/audio_common.h"
@ -352,7 +352,7 @@ void OpenSLESRecorder::AllocateDataBuffers() {
audio_parameters_.GetBytesPerBuffer());
ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
RTC_DCHECK(audio_device_buffer_);
fine_audio_buffer_ = rtc::MakeUnique<FineAudioBuffer>(audio_device_buffer_);
fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
// Allocate queue of audio buffers that stores recorded audio samples.
const int buffer_size_samples =
audio_parameters_.frames_per_buffer() * audio_parameters_.channels();

4
sdk/android/src/jni/pc/datachannel.cc
View File

@ -12,9 +12,9 @@
#include <limits>
#include "absl/memory/memory.h"
#include "api/datachannelinterface.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "sdk/android/generated_peerconnection_jni/jni/DataChannel_jni.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/src/jni/jni_helpers.h"
@ -97,7 +97,7 @@ static jlong JNI_DataChannel_RegisterObserver(
JNIEnv* jni,
const JavaParamRef<jobject>& j_dc,
const JavaParamRef<jobject>& j_observer) {
auto observer = rtc::MakeUnique<DataChannelObserverJni>(jni, j_observer);
auto observer = absl::make_unique<DataChannelObserverJni>(jni, j_observer);
ExtractNativeDC(jni, j_dc)->RegisterObserver(observer.get());
return jlongFromPointer(observer.release());
}

4
sdk/android/src/jni/pc/mediaconstraints.cc
View File

@ -10,7 +10,7 @@
#include "sdk/android/src/jni/pc/mediaconstraints.h"
#include "rtc_base/ptr_util.h"
#include "absl/memory/memory.h"
#include "sdk/android/generated_peerconnection_jni/jni/MediaConstraints_jni.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/src/jni/jni_helpers.h"
@ -60,7 +60,7 @@ class MediaConstraintsJni : public MediaConstraintsInterface {
std::unique_ptr<MediaConstraintsInterface> JavaToNativeMediaConstraints(
JNIEnv* env,
const JavaRef<jobject>& j_constraints) {
return rtc::MakeUnique<MediaConstraintsJni>(env, j_constraints);
return absl::make_unique<MediaConstraintsJni>(env, j_constraints);
}
} // namespace jni

4
sdk/android/src/jni/pc/mediastream.cc
View File

@ -10,7 +10,7 @@
#include "sdk/android/src/jni/pc/mediastream.h"
#include "rtc_base/ptr_util.h"
#include "absl/memory/memory.h"
#include "sdk/android/generated_peerconnection_jni/jni/MediaStream_jni.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/src/jni/jni_helpers.h"
@ -25,7 +25,7 @@ JavaMediaStream::JavaMediaStream(
env,
Java_MediaStream_Constructor(env,
jlongFromPointer(media_stream.get()))),
observer_(rtc::MakeUnique<MediaStreamObserver>(media_stream)) {
observer_(absl::make_unique<MediaStreamObserver>(media_stream)) {
for (rtc::scoped_refptr<AudioTrackInterface> track :
media_stream->GetAudioTracks()) {
Java_MediaStream_addNativeAudioTrack(env, j_media_stream_,

2
sdk/android/src/jni/pc/peerconnection.cc
View File

@ -32,6 +32,7 @@
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "api/mediaconstraintsinterface.h"
#include "api/peerconnectioninterface.h"
#include "api/rtpreceiverinterface.h"
@ -39,7 +40,6 @@
#include "api/rtptransceiverinterface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "sdk/android/generated_peerconnection_jni/jni/PeerConnection_jni.h"
#include "sdk/android/native_api/jni/java_types.h"
#include "sdk/android/src/jni/jni_helpers.h"

4
sdk/android/src/jni/pc/peerconnectionfactory.cc
View File

@ -142,7 +142,7 @@ static void JNI_PeerConnectionFactory_InitializeFieldTrials(
field_trial::InitFieldTrialsFromString(nullptr);
return;
}
field_trials_init_string = rtc::MakeUnique<std::string>(
field_trials_init_string = absl::make_unique<std::string>(
JavaToNativeString(jni, j_trials_init_string));
RTC_LOG(LS_INFO) << "initializeFieldTrials: " << *field_trials_init_string;
field_trial::InitFieldTrialsFromString(field_trials_init_string->c_str());
@ -510,7 +510,7 @@ static void JNI_PeerConnectionFactory_InjectLoggable(
if (jni_log_sink) {
rtc::LogMessage::RemoveLogToStream(jni_log_sink.get());
}
jni_log_sink = rtc::MakeUnique<JNILogSink>(jni, j_logging);
jni_log_sink = absl::make_unique<JNILogSink>(jni, j_logging);
rtc::LogMessage::AddLogToStream(
jni_log_sink.get(), static_cast<rtc::LoggingSeverity>(nativeSeverity));
rtc::LogMessage::LogToDebug(rtc::LS_NONE);