diff options
Diffstat (limited to 'third_party/libwebrtc/modules/async_audio_processing')
4 files changed, 485 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn new file mode 100644 index 0000000000..7a7ca20df1 --- /dev/null +++ b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn @@ -0,0 +1,43 @@ +# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("../../webrtc.gni") + +rtc_library("async_audio_processing") { + sources = [ + "async_audio_processing.cc", + "async_audio_processing.h", + ] + + public = [ "async_audio_processing.h" ] + + deps = [ + "../../api:scoped_refptr", + "../../api:sequence_checker", + "../../api/audio:audio_frame_api", + "../../api/audio:audio_frame_processor", + "../../api/task_queue:task_queue", + "../../rtc_base:checks", + "../../rtc_base:refcount", + "../../rtc_base:rtc_task_queue", + ] +} + +if (rtc_include_tests) { + rtc_library("async_audio_processing_test") { + testonly = true + + sources = [] + + deps = [ + ":async_audio_processing", + "../../api/audio:audio_frame_api", + "../../rtc_base:checks", + ] + } +} diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc new file mode 100644 index 0000000000..19c08dc3e5 --- /dev/null +++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc @@ -0,0 +1,101 @@ + +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#include "modules/async_audio_processing/async_audio_processing.h" + +#include <utility> + +#include "api/audio/audio_frame.h" +#include "api/task_queue/task_queue_factory.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +AsyncAudioProcessing::Factory::~Factory() = default; +AsyncAudioProcessing::Factory::Factory(AudioFrameProcessor& frame_processor, + TaskQueueFactory& task_queue_factory) + : frame_processor_(frame_processor), + task_queue_factory_(task_queue_factory) {} + +AsyncAudioProcessing::Factory::Factory( + std::unique_ptr<AudioFrameProcessor> frame_processor, + TaskQueueFactory& task_queue_factory) + : frame_processor_(*frame_processor), + owned_frame_processor_(std::move(frame_processor)), + task_queue_factory_(task_queue_factory) {} + +std::unique_ptr<AsyncAudioProcessing> +AsyncAudioProcessing::Factory::CreateAsyncAudioProcessing( + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback) { + if (owned_frame_processor_) { + return std::make_unique<AsyncAudioProcessing>( + std::move(owned_frame_processor_), task_queue_factory_, + std::move(on_frame_processed_callback)); + } else { + return std::make_unique<AsyncAudioProcessing>( + frame_processor_, task_queue_factory_, + std::move(on_frame_processed_callback)); + } +} + +AsyncAudioProcessing::~AsyncAudioProcessing() { + if (owned_frame_processor_) { + owned_frame_processor_->SetSink(nullptr); + } else { + frame_processor_.SetSink(nullptr); + } +} + +AsyncAudioProcessing::AsyncAudioProcessing( + AudioFrameProcessor& frame_processor, + TaskQueueFactory& task_queue_factory, + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback) + : on_frame_processed_callback_(std::move(on_frame_processed_callback)), + frame_processor_(frame_processor), + task_queue_(task_queue_factory.CreateTaskQueue( + "AsyncAudioProcessing", + TaskQueueFactory::Priority::NORMAL)) { + frame_processor_.SetSink([this](std::unique_ptr<AudioFrame> frame) { + task_queue_.PostTask([this, frame = std::move(frame)]() mutable { + on_frame_processed_callback_(std::move(frame)); + }); + }); +} + +AsyncAudioProcessing::AsyncAudioProcessing( + std::unique_ptr<AudioFrameProcessor> frame_processor, + TaskQueueFactory& task_queue_factory, + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback) + : on_frame_processed_callback_(std::move(on_frame_processed_callback)), + frame_processor_(*frame_processor), + owned_frame_processor_(std::move(frame_processor)), + task_queue_(task_queue_factory.CreateTaskQueue( + "AsyncAudioProcessing", + TaskQueueFactory::Priority::NORMAL)) { + owned_frame_processor_->SetSink([this](std::unique_ptr<AudioFrame> frame) { + task_queue_.PostTask([this, frame = std::move(frame)]() mutable { + on_frame_processed_callback_(std::move(frame)); + }); + }); +} + +void AsyncAudioProcessing::Process(std::unique_ptr<AudioFrame> frame) { + if (owned_frame_processor_) { + task_queue_.PostTask([this, frame = std::move(frame)]() mutable { + owned_frame_processor_->Process(std::move(frame)); + }); + } else { + task_queue_.PostTask([this, frame = std::move(frame)]() mutable { + frame_processor_.Process(std::move(frame)); + }); + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h new file mode 100644 index 0000000000..f3ed96959b --- /dev/null +++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h @@ -0,0 +1,109 @@ +/* + * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_ +#define MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_ + +#include <memory> + +#include "api/audio/audio_frame_processor.h" +#include "rtc_base/ref_count.h" +#include "rtc_base/task_queue.h" + +namespace webrtc { + +class AudioFrame; +class TaskQueueFactory; + +// Helper class taking care of interactions with AudioFrameProcessor +// in asynchronous manner. Offloads AudioFrameProcessor::Process calls +// to a dedicated task queue. Makes sure that it's always safe for +// AudioFrameProcessor to pass processed frames back to its sink. +class AsyncAudioProcessing final { + public: + // Helper class passing AudioFrameProcessor and TaskQueueFactory into + // AsyncAudioProcessing constructor. + class Factory : public rtc::RefCountInterface { + public: + Factory(const Factory&) = delete; + Factory& operator=(const Factory&) = delete; + + ~Factory(); + Factory(AudioFrameProcessor& frame_processor, + TaskQueueFactory& task_queue_factory); + Factory(std::unique_ptr<AudioFrameProcessor> frame_processor, + TaskQueueFactory& task_queue_factory); + + std::unique_ptr<AsyncAudioProcessing> CreateAsyncAudioProcessing( + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback); + + private: + // TODO(bugs.webrtc.org/15111): + // Remove 'AudioFrameProcessor& frame_processor_' in favour of + // std::unique_ptr in the follow-up. + // While transitioning this API from using AudioFrameProcessor& to using + // std::unique_ptr<AudioFrameProcessor>, we have two member variable both + // referencing the same object. Throughout the lifetime of the Factory + // only one of the variables is used, depending on which constructor was + // called. + AudioFrameProcessor& frame_processor_; + std::unique_ptr<AudioFrameProcessor> owned_frame_processor_; + TaskQueueFactory& task_queue_factory_; + }; + + AsyncAudioProcessing(const AsyncAudioProcessing&) = delete; + AsyncAudioProcessing& operator=(const AsyncAudioProcessing&) = delete; + + ~AsyncAudioProcessing(); + + // Creates AsyncAudioProcessing which will pass audio frames to + // `frame_processor` on `task_queue_` and reply with processed frames passed + // into `on_frame_processed_callback`, which is posted back onto + // `task_queue_`. `task_queue_` is created using the provided + // `task_queue_factory`. + // TODO(bugs.webrtc.org/15111): + // Remove this method in favour of the method taking the + // unique_ptr<AudioFrameProcessor> in the follow-up. + AsyncAudioProcessing( + AudioFrameProcessor& frame_processor, + TaskQueueFactory& task_queue_factory, + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback); + + // Creates AsyncAudioProcessing which will pass audio frames to + // `frame_processor` on `task_queue_` and reply with processed frames passed + // into `on_frame_processed_callback`, which is posted back onto + // `task_queue_`. `task_queue_` is created using the provided + // `task_queue_factory`. + AsyncAudioProcessing( + std::unique_ptr<AudioFrameProcessor> frame_processor, + TaskQueueFactory& task_queue_factory, + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback); + + // Accepts `frame` for asynchronous processing. Thread-safe. + void Process(std::unique_ptr<AudioFrame> frame); + + private: + AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback_; + // TODO(bugs.webrtc.org/15111): + // Remove 'AudioFrameProcessor& frame_processor_' in favour of + // std::unique_ptr in the follow-up. + // While transitioning this API from using AudioFrameProcessor& to using + // std::unique_ptr<AudioFrameProcessor>, we have two member variable both + // referencing the same object. Throughout the lifetime of the Factory + // only one of the variables is used, depending on which constructor was + // called. + AudioFrameProcessor& frame_processor_; + std::unique_ptr<AudioFrameProcessor> owned_frame_processor_; + rtc::TaskQueue task_queue_; +}; + +} // namespace webrtc + +#endif // MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_ diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build new file mode 100644 index 0000000000..347559a342 --- /dev/null +++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build @@ -0,0 +1,232 @@ +# This Source Code Form is subject to the terms of the Mozilla Public +# License, v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. + + + ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ### + ### DO NOT edit it by hand. ### + +COMPILE_FLAGS["OS_INCLUDES"] = [] +AllowCompilerWarnings() + +DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1" +DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True +DEFINES["RTC_ENABLE_VP9"] = True +DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0" +DEFINES["WEBRTC_LIBRARY_IMPL"] = True +DEFINES["WEBRTC_MOZILLA_BUILD"] = True +DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0" +DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0" + +FINAL_LIBRARY = "webrtc" + + +LOCAL_INCLUDES += [ + "!/ipc/ipdl/_ipdlheaders", + "!/third_party/libwebrtc/gen", + "/ipc/chromium/src", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc" +] + +if not CONFIG["MOZ_DEBUG"]: + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0" + DEFINES["NDEBUG"] = True + DEFINES["NVALGRIND"] = True + +if CONFIG["MOZ_DEBUG"] == "1": + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1" + +if CONFIG["OS_TARGET"] == "Android": + + DEFINES["ANDROID"] = True + DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1" + DEFINES["HAVE_SYS_UIO_H"] = True + DEFINES["WEBRTC_ANDROID"] = True + DEFINES["WEBRTC_ANDROID_OPENSLES"] = True + DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True + DEFINES["WEBRTC_LINUX"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_GNU_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + + OS_LIBS += [ + "log" + ] + +if CONFIG["OS_TARGET"] == "Darwin": + + DEFINES["WEBRTC_MAC"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True + DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0" + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "Linux": + + DEFINES["USE_AURA"] = "1" + DEFINES["USE_GLIB"] = "1" + DEFINES["USE_NSS_CERTS"] = "1" + DEFINES["USE_OZONE"] = "1" + DEFINES["USE_UDEV"] = True + DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True + DEFINES["WEBRTC_LINUX"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_FILE_OFFSET_BITS"] = "64" + DEFINES["_LARGEFILE64_SOURCE"] = True + DEFINES["_LARGEFILE_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "OpenBSD": + + DEFINES["USE_GLIB"] = "1" + DEFINES["USE_OZONE"] = "1" + DEFINES["USE_X11"] = "1" + DEFINES["WEBRTC_BSD"] = True + DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True + DEFINES["WEBRTC_POSIX"] = True + DEFINES["_FILE_OFFSET_BITS"] = "64" + DEFINES["_LARGEFILE64_SOURCE"] = True + DEFINES["_LARGEFILE_SOURCE"] = True + DEFINES["__STDC_CONSTANT_MACROS"] = True + DEFINES["__STDC_FORMAT_MACROS"] = True + +if CONFIG["OS_TARGET"] == "WINNT": + + DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True + DEFINES["NOMINMAX"] = True + DEFINES["NTDDI_VERSION"] = "0x0A000000" + DEFINES["PSAPI_VERSION"] = "2" + DEFINES["RTC_ENABLE_WIN_WGC"] = True + DEFINES["UNICODE"] = True + DEFINES["USE_AURA"] = "1" + DEFINES["WEBRTC_WIN"] = True + DEFINES["WIN32"] = True + DEFINES["WIN32_LEAN_AND_MEAN"] = True + DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP" + DEFINES["WINVER"] = "0x0A00" + DEFINES["_ATL_NO_OPENGL"] = True + DEFINES["_CRT_RAND_S"] = True + DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True + DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True + DEFINES["_HAS_EXCEPTIONS"] = "0" + DEFINES["_HAS_NODISCARD"] = True + DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True + DEFINES["_SECURE_ATL"] = True + DEFINES["_UNICODE"] = True + DEFINES["_WIN32_WINNT"] = "0x0A00" + DEFINES["_WINDOWS"] = True + DEFINES["__STD_C"] = True + + OS_LIBS += [ + "crypt32", + "iphlpapi", + "secur32", + "winmm" + ] + +if CONFIG["TARGET_CPU"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["TARGET_CPU"] == "arm": + + CXXFLAGS += [ + "-mfpu=neon" + ] + + DEFINES["WEBRTC_ARCH_ARM"] = True + DEFINES["WEBRTC_ARCH_ARM_V7"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["TARGET_CPU"] == "mips32": + + DEFINES["MIPS32_LE"] = True + DEFINES["MIPS_FPU_LE"] = True + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["TARGET_CPU"] == "mips64": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["TARGET_CPU"] == "x86": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["TARGET_CPU"] == "x86_64": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD": + + DEFINES["_DEBUG"] = True + +if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT": + + DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0" + +if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["USE_X11"] = "1" + +if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm": + + OS_LIBS += [ + "android_support", + "unwind" + ] + +if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86": + + CXXFLAGS += [ + "-msse2" + ] + + OS_LIBS += [ + "android_support" + ] + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86": + + CXXFLAGS += [ + "-msse2" + ] + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64": + + DEFINES["_GNU_SOURCE"] = True + +Library("async_audio_processing_gn") |