Propagate field trials into AudioDeviceIOS
Bug: webrtc:413413572
Change-Id: I96fb90f4bf0304045fc3141c96955fd3f2df043a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/389220
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44619}
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index da98848..585d637 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -102,7 +102,7 @@
}
// Create the platform-dependent implementation.
- if (audio_device->CreatePlatformSpecificObjects() == -1) {
+ if (audio_device->CreatePlatformSpecificObjects(env) == -1) {
return nullptr;
}
@@ -166,7 +166,8 @@
return 0;
}
-int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
+int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects(
+ [[maybe_unused]] const Environment& env) {
RTC_LOG(LS_INFO) << __FUNCTION__;
if (audio_device_ != nullptr) {
RTC_LOG(LS_INFO) << "Reusing provided audio device";
@@ -241,10 +242,11 @@
// iOS ADM implementation.
#if defined(WEBRTC_IOS)
if (audio_layer == kPlatformDefaultAudio) {
- audio_device_.reset(new ios_adm::AudioDeviceIOS(
+ audio_device_ = std::make_unique<ios_adm::AudioDeviceIOS>(
+ env,
/*bypass_voice_processing=*/false,
/*muted_speech_event_handler=*/nullptr,
- /*render_error_handler=*/nullptr));
+ /*render_error_handler=*/nullptr);
RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
}
// END #if defined(WEBRTC_IOS)
diff --git a/modules/audio_device/audio_device_impl.h b/modules/audio_device/audio_device_impl.h
index 609daf5..c7a229d 100644
--- a/modules/audio_device/audio_device_impl.h
+++ b/modules/audio_device/audio_device_impl.h
@@ -61,7 +61,7 @@
~AudioDeviceModuleImpl() override;
int32_t CheckPlatform();
- int32_t CreatePlatformSpecificObjects();
+ int32_t CreatePlatformSpecificObjects(const Environment& env);
int32_t AttachAudioBuffer();
// Retrieve the currently utilized audio layer
diff --git a/modules/audio_device/include/test_audio_device.cc b/modules/audio_device/include/test_audio_device.cc
index f6d5386..3b58ff2 100644
--- a/modules/audio_device/include/test_audio_device.cc
+++ b/modules/audio_device/include/test_audio_device.cc
@@ -453,7 +453,7 @@
}
// Create the platform-dependent implementation.
- if (audio_device->CreatePlatformSpecificObjects() == -1) {
+ if (audio_device->CreatePlatformSpecificObjects(env) == -1) {
return nullptr;
}
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index c0c8eb4..c443527 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -231,6 +231,10 @@
rtc_library("native_api_audio_device_module") {
visibility = [ "*" ]
+ # TODO: bugs.webrtc.org/413413572 - Remove the poison when users of this
+ # api provide Environment explicitly.
+ allow_poison = [ "environment_construction" ]
+
sources = [
"objc/native/api/audio_device_module.h",
"objc/native/api/audio_device_module.mm",
@@ -240,7 +244,10 @@
":audio_device",
":audio_device_module_error_handler",
"../api:make_ref_counted",
+ "../api:scoped_refptr",
"../api/audio:audio_device",
+ "../api/environment",
+ "../api/environment:environment_factory",
"../modules/audio_device:audio_device_generic",
"../rtc_base:checks",
"../rtc_base:logging",
@@ -306,8 +313,8 @@
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_device",
+ "../api/environment",
"../api/task_queue",
- "../api/task_queue:default_task_queue_factory",
"../api/task_queue:pending_task_safety_flag",
"../modules/audio_device:audio_device_buffer",
"../modules/audio_device:audio_device_config",
@@ -485,7 +492,6 @@
rtc_library("audio_device_objc") {
visibility = [ "*" ]
- allow_poison = [ "environment_construction" ]
sources = [
"objc/native/src/objc_audio_device.h",
"objc/native/src/objc_audio_device.mm",
@@ -501,8 +507,8 @@
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/audio:audio_device",
+ "../api/environment",
"../api/task_queue",
- "../api/task_queue:default_task_queue_factory",
"../modules/audio_device:audio_device_buffer",
"../rtc_base:buffer",
"../rtc_base:checks",
@@ -528,7 +534,6 @@
rtc_library("objc_audio_device_module") {
visibility = [ "*" ]
- allow_poison = [ "environment_construction" ]
sources = [
"objc/native/api/objc_audio_device_module.h",
"objc/native/api/objc_audio_device_module.mm",
@@ -538,7 +543,9 @@
":audio_device_api_objc",
":audio_device_objc",
"../api:make_ref_counted",
+ "../api:scoped_refptr",
"../api/audio:audio_device",
+ "../api/environment",
"../rtc_base:logging",
]
if (is_mac) {
@@ -1209,6 +1216,7 @@
"../api/audio:audio_processing",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/environment",
"../api/environment:environment_factory",
"../api/task_queue:default_task_queue_factory",
"../api/video:video_frame",
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
index 250ba40..5df145e 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -62,14 +62,6 @@
@synthesize nativeFactory = _nativeFactory;
-- (webrtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
-#if defined(WEBRTC_IOS)
- return webrtc::CreateAudioDeviceModule();
-#else
- return nullptr;
-#endif
-}
-
- (instancetype)init {
webrtc::PeerConnectionFactoryDependencies dependencies;
dependencies.audio_encoder_factory =
@@ -80,7 +72,10 @@
[[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]);
dependencies.video_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(
[[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]);
- dependencies.adm = [self audioDeviceModule];
+ dependencies.env = webrtc::CreateEnvironment();
+#ifdef WEBRTC_IOS
+ dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env);
+#endif
return [self initWithMediaAndDependencies:dependencies];
}
@@ -105,6 +100,7 @@
return [self initWithNoMedia];
#else
webrtc::PeerConnectionFactoryDependencies dependencies;
+ dependencies.env = webrtc::CreateEnvironment();
dependencies.audio_encoder_factory =
webrtc::CreateBuiltinAudioEncoderFactory();
dependencies.audio_decoder_factory =
@@ -118,9 +114,12 @@
webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
}
if (audioDevice) {
- dependencies.adm = webrtc::CreateAudioDeviceModule(audioDevice);
+ dependencies.adm =
+ webrtc::CreateAudioDeviceModule(*dependencies.env, audioDevice);
+#ifdef WEBRTC_IOS
} else {
- dependencies.adm = [self audioDeviceModule];
+ dependencies.adm = webrtc::CreateAudioDeviceModule(*dependencies.env);
+#endif
}
return [self initWithMediaAndDependencies:dependencies];
#endif
diff --git a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
index e5eeba5..3e18fd1 100644
--- a/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
+++ b/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
@@ -19,6 +19,7 @@
#include "sdk/objc/native/api/video_encoder_factory.h"
#if defined(WEBRTC_IOS)
+#include "api/environment/environment_factory.h"
#import "sdk/objc/native/api/audio_device_module.h"
#endif
@@ -42,7 +43,8 @@
[builder setVideoDecoderFactory:std::move(videoDecoderFactory)];
#if defined(WEBRTC_IOS)
- [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule()];
+ [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule(
+ webrtc::CreateEnvironment())];
#endif
return builder;
}
diff --git a/sdk/objc/native/api/audio_device_module.h b/sdk/objc/native/api/audio_device_module.h
index 34f99e8..5a8944c 100644
--- a/sdk/objc/native/api/audio_device_module.h
+++ b/sdk/objc/native/api/audio_device_module.h
@@ -14,6 +14,8 @@
#include <memory>
#include "api/audio/audio_device.h"
+#include "api/environment/environment.h"
+#include "api/scoped_refptr.h"
#include "sdk/objc/native/api/audio_device_module_error_handler.h"
namespace webrtc {
@@ -23,20 +25,26 @@
// Warning: Setting `bypass_voice_processing` will have unpredictable
// consequences for the audio path in the device. It is not advisable to use in
// most scenarios.
-webrtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+ const Environment& env,
bool bypass_voice_processing = false);
-// If `muted_speech_event_handler` is exist, audio unit will catch speech
-// activity while muted.
-webrtc::scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
- AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
+[[deprecated("Pass `env` explicitly instead of relying on the default")]]
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
bool bypass_voice_processing = false);
// If `muted_speech_event_handler` is exist, audio unit will catch speech
// activity while muted.
// Provide `error_handler` to receive callbacks on errors such as microphone
// init failed or playout start failied.
-webrtc::scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
+scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
+ const Environment& env,
+ AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
+ ADMErrorHandler error_handler,
+ bool bypass_voice_processing = false);
+
+[[deprecated("Pass `env` explicitly instead of relying on the default")]]
+scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
ADMErrorHandler error_handler,
bool bypass_voice_processing = false);
diff --git a/sdk/objc/native/api/audio_device_module.mm b/sdk/objc/native/api/audio_device_module.mm
index 898886b..8325179 100644
--- a/sdk/objc/native/api/audio_device_module.mm
+++ b/sdk/objc/native/api/audio_device_module.mm
@@ -10,49 +10,57 @@
#include "audio_device_module.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
#include "api/make_ref_counted.h"
#include "rtc_base/logging.h"
-
#include "sdk/objc/native/src/audio/audio_device_module_ios.h"
+#ifndef WEBRTC_IOS
+#error This file shouldn't be compiled on platforms other than IOS.
+#endif
+
namespace webrtc {
-webrtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
- bool bypass_voice_processing) {
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+ const Environment& env, bool bypass_voice_processing) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
-#if defined(WEBRTC_IOS)
- return webrtc::make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
+ return make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
+ env,
bypass_voice_processing,
/*muted_speech_event_handler=*/nullptr,
/*error_handler=*/nullptr);
-#else
- RTC_LOG(LS_ERROR)
- << "current platform is not supported => this module will self destruct!";
- return nullptr;
-#endif
}
-webrtc::scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
- AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
bool bypass_voice_processing) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
- return CreateMutedDetectAudioDeviceModule(muted_speech_event_handler,
- /*error_handler=*/nullptr,
- bypass_voice_processing);
+ return make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
+ CreateEnvironment(),
+ bypass_voice_processing,
+ /*muted_speech_event_handler=*/nullptr,
+ /*error_handler=*/nullptr);
}
-webrtc::scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
+scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
+ const Environment& env,
AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
ADMErrorHandler error_handler,
bool bypass_voice_processing) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
-#if defined(WEBRTC_IOS)
- return webrtc::make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
- bypass_voice_processing, muted_speech_event_handler, error_handler);
-#else
- RTC_LOG(LS_ERROR)
- << "current platform is not supported => this module will self destruct!";
- return nullptr;
-#endif
+ return make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
+ env, bypass_voice_processing, muted_speech_event_handler, error_handler);
+}
+
+scoped_refptr<AudioDeviceModule> CreateMutedDetectAudioDeviceModule(
+ AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
+ ADMErrorHandler error_handler,
+ bool bypass_voice_processing) {
+ RTC_DLOG(LS_INFO) << __FUNCTION__;
+ return make_ref_counted<ios_adm::AudioDeviceModuleIOS>(
+ CreateEnvironment(),
+ bypass_voice_processing,
+ muted_speech_event_handler,
+ error_handler);
}
} // namespace webrtc
diff --git a/sdk/objc/native/api/objc_audio_device_module.h b/sdk/objc/native/api/objc_audio_device_module.h
index a4387df..be1b344 100644
--- a/sdk/objc/native/api/objc_audio_device_module.h
+++ b/sdk/objc/native/api/objc_audio_device_module.h
@@ -12,11 +12,14 @@
#define SDK_OBJC_NATIVE_API_OBJC_AUDIO_DEVICE_MODULE_H_
#include "api/audio/audio_device.h"
+#include "api/environment/environment.h"
+#include "api/scoped_refptr.h"
#import "components/audio/RTCAudioDevice.h"
namespace webrtc {
-webrtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+ const Environment& env,
id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
} // namespace webrtc
diff --git a/sdk/objc/native/api/objc_audio_device_module.mm b/sdk/objc/native/api/objc_audio_device_module.mm
index cedb38f..3eb78ec 100644
--- a/sdk/objc/native/api/objc_audio_device_module.mm
+++ b/sdk/objc/native/api/objc_audio_device_module.mm
@@ -10,6 +10,7 @@
#include "objc_audio_device_module.h"
+#include "api/environment/environment.h"
#include "api/make_ref_counted.h"
#include "rtc_base/logging.h"
@@ -17,11 +18,10 @@
namespace webrtc {
-webrtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
- id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
+scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+ const Environment& env, id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
RTC_DLOG(LS_INFO) << __FUNCTION__;
- return webrtc::make_ref_counted<objc_adm::ObjCAudioDeviceModule>(
- audio_device);
+ return make_ref_counted<objc_adm::ObjCAudioDeviceModule>(env, audio_device);
}
} // namespace webrtc
diff --git a/sdk/objc/native/src/audio/audio_device_ios.h b/sdk/objc/native/src/audio/audio_device_ios.h
index cc70ee7..7dcffb0 100644
--- a/sdk/objc/native/src/audio/audio_device_ios.h
+++ b/sdk/objc/native/src/audio/audio_device_ios.h
@@ -14,6 +14,7 @@
#include <atomic>
#include <memory>
+#include "api/environment/environment.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
@@ -56,6 +57,7 @@
public VoiceProcessingAudioUnitObserver {
public:
explicit AudioDeviceIOS(
+ const Environment& env,
bool bypass_voice_processing,
AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
AudioDeviceIOSRenderErrorHandler render_error_handler);
@@ -220,6 +222,8 @@
// Resets thread-checkers before a call is restarted.
void PrepareForNewStart();
+ const Environment env_;
+
// Determines whether voice processing should be enabled or disabled.
const bool bypass_voice_processing_;
diff --git a/sdk/objc/native/src/audio/audio_device_ios.mm b/sdk/objc/native/src/audio/audio_device_ios.mm
index c7e8b62..53dec77 100644
--- a/sdk/objc/native/src/audio/audio_device_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_ios.mm
@@ -17,6 +17,7 @@
#include <cmath>
#include "api/array_view.h"
+#include "api/environment/environment.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "helpers.h"
#include "modules/audio_device/fine_audio_buffer.h"
@@ -25,7 +26,6 @@
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
-#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#import "base/RTCLogging.h"
@@ -96,10 +96,12 @@
#endif // !defined(NDEBUG)
AudioDeviceIOS::AudioDeviceIOS(
+ const Environment& env,
bool bypass_voice_processing,
AudioDeviceModule::MutedSpeechEventHandler muted_speech_event_handler,
AudioDeviceIOSRenderErrorHandler render_error_handler)
- : bypass_voice_processing_(bypass_voice_processing),
+ : env_(env),
+ bypass_voice_processing_(bypass_voice_processing),
muted_speech_event_handler_(muted_speech_event_handler),
render_error_handler_(render_error_handler),
disregard_next_render_error_(false),
@@ -579,7 +581,7 @@
is_interrupted_);
is_interrupted_ = false;
if (!audio_unit_) return;
- if (webrtc::field_trial::IsEnabled("WebRTC-Audio-iOS-Holding")) {
+ if (env_.field_trials().IsEnabled("WebRTC-Audio-iOS-Holding")) {
// Work around an issue where audio does not restart properly after an
// interruption by restarting the audio unit when the interruption ends.
if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
diff --git a/sdk/objc/native/src/audio/audio_device_module_ios.h b/sdk/objc/native/src/audio/audio_device_module_ios.h
index 394e1ff..5ff5062 100644
--- a/sdk/objc/native/src/audio/audio_device_module_ios.h
+++ b/sdk/objc/native/src/audio/audio_device_module_ios.h
@@ -14,7 +14,6 @@
#include <memory>
#include "api/audio/audio_device.h"
-#include "api/task_queue/task_queue_factory.h"
#include "audio_device_ios.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "rtc_base/checks.h"
@@ -31,6 +30,7 @@
int32_t AttachAudioBuffer();
explicit AudioDeviceModuleIOS(
+ const Environment& env,
bool bypass_voice_processing,
MutedSpeechEventHandler muted_speech_event_handler,
ADMErrorHandler error_handler);
@@ -137,11 +137,12 @@
#endif // WEBRTC_IOS
private:
void ReportError(ADMError error) const;
+
+ const Environment env_;
const bool bypass_voice_processing_;
MutedSpeechEventHandler muted_speech_event_handler_;
ADMErrorHandler error_handler_;
bool initialized_ = false;
- const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
std::unique_ptr<AudioDeviceIOS> audio_device_;
std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
};
diff --git a/sdk/objc/native/src/audio/audio_device_module_ios.mm b/sdk/objc/native/src/audio/audio_device_module_ios.mm
index 3b338f2..7420d05 100644
--- a/sdk/objc/native/src/audio/audio_device_module_ios.mm
+++ b/sdk/objc/native/src/audio/audio_device_module_ios.mm
@@ -10,7 +10,9 @@
#include "audio_device_module_ios.h"
-#include "api/task_queue/default_task_queue_factory.h"
+#include <memory>
+
+#include "api/environment/environment.h"
#include "modules/audio_device/audio_device_config.h"
#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/checks.h"
@@ -42,13 +44,14 @@
namespace ios_adm {
AudioDeviceModuleIOS::AudioDeviceModuleIOS(
+ const Environment& env,
bool bypass_voice_processing,
MutedSpeechEventHandler muted_speech_event_handler,
ADMErrorHandler error_handler)
- : bypass_voice_processing_(bypass_voice_processing),
+ : env_(env),
+ bypass_voice_processing_(bypass_voice_processing),
muted_speech_event_handler_(muted_speech_event_handler),
- error_handler_(error_handler),
- task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+ error_handler_(error_handler) {
RTC_LOG(LS_INFO) << "current platform is IOS";
RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
}
@@ -87,10 +90,13 @@
AudioDeviceIOSRenderErrorHandler error_handler = ^(OSStatus error) {
ReportError(kRecordingDeviceFailed);
};
- audio_device_buffer_.reset(
- new webrtc::AudioDeviceBuffer(task_queue_factory_.get()));
- audio_device_.reset(new ios_adm::AudioDeviceIOS(
- bypass_voice_processing_, muted_speech_event_handler_, error_handler));
+ audio_device_buffer_ =
+ std::make_unique<AudioDeviceBuffer>(&env_.task_queue_factory());
+ audio_device_ =
+ std::make_unique<ios_adm::AudioDeviceIOS>(env_,
+ bypass_voice_processing_,
+ muted_speech_event_handler_,
+ error_handler);
RTC_CHECK(audio_device_);
this->AttachAudioBuffer();
diff --git a/sdk/objc/native/src/objc_audio_device.h b/sdk/objc/native/src/objc_audio_device.h
index 27eb0af..d8edde6 100644
--- a/sdk/objc/native/src/objc_audio_device.h
+++ b/sdk/objc/native/src/objc_audio_device.h
@@ -16,6 +16,7 @@
#import "components/audio/RTCAudioDevice.h"
#include "api/audio/audio_device.h"
+#include "api/environment/environment.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "rtc_base/thread.h"
@@ -30,7 +31,7 @@
class ObjCAudioDeviceModule : public AudioDeviceModule {
public:
explicit ObjCAudioDeviceModule(
- id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
+ const Environment& env, id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
~ObjCAudioDeviceModule() override;
// Retrieve the currently utilized audio layer
@@ -219,7 +220,7 @@
private:
id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device_;
- const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+ const Environment env_;
// AudioDeviceBuffer is a buffer to consume audio recorded by `RTCAudioDevice`
// and provide audio to be played via `RTCAudioDevice`.
diff --git a/sdk/objc/native/src/objc_audio_device.mm b/sdk/objc/native/src/objc_audio_device.mm
index da595ba..d17dd63 100644
--- a/sdk/objc/native/src/objc_audio_device.mm
+++ b/sdk/objc/native/src/objc_audio_device.mm
@@ -9,12 +9,12 @@
*/
#include "objc_audio_device.h"
-#include "objc_audio_device_delegate.h"
+
+#include <memory>
#import "components/audio/RTCAudioDevice.h"
#include "modules/audio_device/fine_audio_buffer.h"
-
-#include "api/task_queue/default_task_queue_factory.h"
+#include "objc_audio_device_delegate.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/time_utils.h"
@@ -49,9 +49,8 @@
namespace objc_adm {
ObjCAudioDeviceModule::ObjCAudioDeviceModule(
- id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device)
- : audio_device_(audio_device),
- task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+ const Environment& env, id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device)
+ : audio_device_(audio_device), env_(env) {
RTC_DLOG_F(LS_VERBOSE) << "";
RTC_DCHECK(audio_device_);
thread_checker_.Detach();
@@ -82,8 +81,8 @@
io_record_thread_checker_.Detach();
thread_ = Thread::Current();
- audio_device_buffer_.reset(
- new webrtc::AudioDeviceBuffer(task_queue_factory_.get()));
+ audio_device_buffer_ =
+ std::make_unique<webrtc::AudioDeviceBuffer>(&env_.task_queue_factory());
if (![audio_device_ isInitialized]) {
if (audio_device_delegate_ == nil) {
diff --git a/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm b/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
index e0ca410..b87460c 100644
--- a/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
+++ b/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
@@ -16,6 +16,7 @@
#import "sdk/objc/native/api/audio_device_module.h"
#endif
+#include "api/environment/environment_factory.h"
#include "api/scoped_refptr.h"
typedef int32_t (^NeedMorePlayDataBlock)(const size_t nSamples,
@@ -156,7 +157,8 @@
_testEnabled = true;
#endif
- audioDeviceModule = webrtc::CreateAudioDeviceModule();
+ audioDeviceModule =
+ webrtc::CreateAudioDeviceModule(webrtc::CreateEnvironment());
XCTAssertEqual(0, audioDeviceModule->Init());
XCTAssertEqual(
0, audioDeviceModule->GetPlayoutAudioParameters(&playoutParameters));
@@ -264,7 +266,7 @@
// Create and initialize a second/extra ADM instance. The default ADM is
// created by the test harness.
webrtc::scoped_refptr<webrtc::AudioDeviceModule> secondAudioDeviceModule =
- webrtc::CreateAudioDeviceModule();
+ webrtc::CreateAudioDeviceModule(webrtc::CreateEnvironment());
XCTAssertNotEqual(secondAudioDeviceModule.get(), nullptr);
XCTAssertEqual(0, secondAudioDeviceModule->Init());
diff --git a/sdk/objc/unittests/RTCAudioDevice_xctest.mm b/sdk/objc/unittests/RTCAudioDevice_xctest.mm
index 5bdb90c..f117832 100644
--- a/sdk/objc/unittests/RTCAudioDevice_xctest.mm
+++ b/sdk/objc/unittests/RTCAudioDevice_xctest.mm
@@ -12,6 +12,8 @@
#include <stdlib.h>
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
@@ -45,8 +47,10 @@
_testEnabled = true;
#endif
- _audioDeviceModule = webrtc::CreateAudioDeviceModule();
+ webrtc::Environment env = webrtc::CreateEnvironment();
+ _audioDeviceModule = webrtc::CreateAudioDeviceModule(env);
_audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS(
+ env,
/*bypass_voice_processing=*/false,
/*muted_speech_event_handler=*/nullptr,
/*render_error_handler=*/nullptr));
@@ -149,6 +153,7 @@
};
_audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS(
+ webrtc::CreateEnvironment(),
/*bypass_voice_processing=*/false,
/*muted_speech_event_handler=*/muted_speech_event_handler,
/*render_error_handler=*/nullptr));
@@ -169,6 +174,7 @@
};
_audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS(
+ webrtc::CreateEnvironment(),
/*bypass_voice_processing=*/false,
/*muted_speech_event_handler=*/muted_speech_event_handler,
/*render_error_handler=*/nullptr));
