From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/audio/BUILD.gn | 93 +++++++ third_party/libwebrtc/api/audio/OWNERS | 2 + .../libwebrtc/api/audio/aec3_config_gn/moz.build | 225 ++++++++++++++++ .../libwebrtc/api/audio/aec3_factory_gn/moz.build | 237 +++++++++++++++++ third_party/libwebrtc/api/audio/audio_frame.cc | 140 ++++++++++ third_party/libwebrtc/api/audio/audio_frame.h | 173 +++++++++++++ .../api/audio/audio_frame_api_gn/moz.build | 233 +++++++++++++++++ .../libwebrtc/api/audio/audio_frame_processor.h | 43 ++++ .../api/audio/audio_frame_processor_gn/moz.build | 205 +++++++++++++++ third_party/libwebrtc/api/audio/audio_mixer.h | 80 ++++++ .../api/audio/audio_mixer_api_gn/moz.build | 216 ++++++++++++++++ third_party/libwebrtc/api/audio/channel_layout.cc | 282 +++++++++++++++++++++ third_party/libwebrtc/api/audio/channel_layout.h | 165 ++++++++++++ .../libwebrtc/api/audio/echo_canceller3_config.cc | 278 ++++++++++++++++++++ .../libwebrtc/api/audio/echo_canceller3_config.h | 250 ++++++++++++++++++ .../libwebrtc/api/audio/echo_canceller3_factory.cc | 32 +++ .../libwebrtc/api/audio/echo_canceller3_factory.h | 41 +++ third_party/libwebrtc/api/audio/echo_control.h | 75 ++++++ .../libwebrtc/api/audio/echo_control_gn/moz.build | 209 +++++++++++++++ .../libwebrtc/api/audio/echo_detector_creator.cc | 21 ++ .../libwebrtc/api/audio/echo_detector_creator.h | 26 ++ third_party/libwebrtc/api/audio/test/BUILD.gn | 29 +++ .../api/audio/test/audio_frame_unittest.cc | 136 ++++++++++ .../audio/test/echo_canceller3_config_unittest.cc | 46 ++++ 24 files changed, 3237 insertions(+) create mode 100644 third_party/libwebrtc/api/audio/BUILD.gn create mode 100644 third_party/libwebrtc/api/audio/OWNERS create mode 100644 third_party/libwebrtc/api/audio/aec3_config_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/aec3_factory_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/audio_frame.cc create mode 100644 third_party/libwebrtc/api/audio/audio_frame.h create mode 100644 third_party/libwebrtc/api/audio/audio_frame_api_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/audio_frame_processor.h create mode 100644 third_party/libwebrtc/api/audio/audio_frame_processor_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/audio_mixer.h create mode 100644 third_party/libwebrtc/api/audio/audio_mixer_api_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/channel_layout.cc create mode 100644 third_party/libwebrtc/api/audio/channel_layout.h create mode 100644 third_party/libwebrtc/api/audio/echo_canceller3_config.cc create mode 100644 third_party/libwebrtc/api/audio/echo_canceller3_config.h create mode 100644 third_party/libwebrtc/api/audio/echo_canceller3_factory.cc create mode 100644 third_party/libwebrtc/api/audio/echo_canceller3_factory.h create mode 100644 third_party/libwebrtc/api/audio/echo_control.h create mode 100644 third_party/libwebrtc/api/audio/echo_control_gn/moz.build create mode 100644 third_party/libwebrtc/api/audio/echo_detector_creator.cc create mode 100644 third_party/libwebrtc/api/audio/echo_detector_creator.h create mode 100644 third_party/libwebrtc/api/audio/test/BUILD.gn create mode 100644 third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc create mode 100644 third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc (limited to 'third_party/libwebrtc/api/audio') diff --git a/third_party/libwebrtc/api/audio/BUILD.gn b/third_party/libwebrtc/api/audio/BUILD.gn new file mode 100644 index 0000000000..0ecab64086 --- /dev/null +++ b/third_party/libwebrtc/api/audio/BUILD.gn @@ -0,0 +1,93 @@ +# Copyright (c) 2018 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("audio_frame_api") { + visibility = [ "*" ] + sources = [ + "audio_frame.cc", + "audio_frame.h", + "channel_layout.cc", + "channel_layout.h", + ] + + deps = [ + "..:rtp_packet_info", + "../../rtc_base:checks", + "../../rtc_base:logging", + "../../rtc_base:macromagic", + "../../rtc_base:timeutils", + ] +} + +rtc_source_set("audio_frame_processor") { + visibility = [ "*" ] + sources = [ "audio_frame_processor.h" ] +} + +rtc_source_set("audio_mixer_api") { + visibility = [ "*" ] + sources = [ "audio_mixer.h" ] + + deps = [ + ":audio_frame_api", + "..:make_ref_counted", + "../../rtc_base:refcount", + ] +} + +rtc_library("aec3_config") { + visibility = [ "*" ] + sources = [ + "echo_canceller3_config.cc", + "echo_canceller3_config.h", + ] + deps = [ + "../../rtc_base:checks", + "../../rtc_base:safe_minmax", + "../../rtc_base/system:rtc_export", + ] +} + +rtc_library("aec3_factory") { + visibility = [ "*" ] + configs += [ "../../modules/audio_processing:apm_debug_dump" ] + sources = [ + "echo_canceller3_factory.cc", + "echo_canceller3_factory.h", + ] + + deps = [ + ":aec3_config", + ":echo_control", + "../../modules/audio_processing/aec3", + "../../rtc_base/system:rtc_export", + ] +} + +rtc_source_set("echo_control") { + visibility = [ "*" ] + sources = [ "echo_control.h" ] + deps = [ "../../rtc_base:checks" ] +} + +rtc_source_set("echo_detector_creator") { + visibility = [ "*" ] + allow_poison = [ "default_echo_detector" ] + sources = [ + "echo_detector_creator.cc", + "echo_detector_creator.h", + ] + deps = [ + "..:make_ref_counted", + "../../api:scoped_refptr", + "../../modules/audio_processing:api", + "../../modules/audio_processing:residual_echo_detector", + ] +} diff --git a/third_party/libwebrtc/api/audio/OWNERS b/third_party/libwebrtc/api/audio/OWNERS new file mode 100644 index 0000000000..bb499b450f --- /dev/null +++ b/third_party/libwebrtc/api/audio/OWNERS @@ -0,0 +1,2 @@ +gustaf@webrtc.org +peah@webrtc.org diff --git a/third_party/libwebrtc/api/audio/aec3_config_gn/moz.build b/third_party/libwebrtc/api/audio/aec3_config_gn/moz.build new file mode 100644 index 0000000000..4b96910919 --- /dev/null +++ b/third_party/libwebrtc/api/audio/aec3_config_gn/moz.build @@ -0,0 +1,225 @@ +# 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/api/audio/echo_canceller3_config.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 + +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("aec3_config_gn") diff --git a/third_party/libwebrtc/api/audio/aec3_factory_gn/moz.build b/third_party/libwebrtc/api/audio/aec3_factory_gn/moz.build new file mode 100644 index 0000000000..de044719cc --- /dev/null +++ b/third_party/libwebrtc/api/audio/aec3_factory_gn/moz.build @@ -0,0 +1,237 @@ +# 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_APM_DEBUG_DUMP"] = "1" +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/api/audio/echo_canceller3_factory.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 + + OS_LIBS += [ + "rt" + ] + +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("aec3_factory_gn") diff --git a/third_party/libwebrtc/api/audio/audio_frame.cc b/third_party/libwebrtc/api/audio/audio_frame.cc new file mode 100644 index 0000000000..3e12006386 --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame.cc @@ -0,0 +1,140 @@ +/* + * Copyright (c) 2018 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 "api/audio/audio_frame.h" + +#include + +#include "rtc_base/checks.h" +#include "rtc_base/time_utils.h" + +namespace webrtc { + +AudioFrame::AudioFrame() { + // Visual Studio doesn't like this in the class definition. + static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes"); +} + +void AudioFrame::Reset() { + ResetWithoutMuting(); + muted_ = true; +} + +void AudioFrame::ResetWithoutMuting() { + // TODO(wu): Zero is a valid value for `timestamp_`. We should initialize + // to an invalid value, or add a new member to indicate invalidity. + timestamp_ = 0; + elapsed_time_ms_ = -1; + ntp_time_ms_ = -1; + samples_per_channel_ = 0; + sample_rate_hz_ = 0; + num_channels_ = 0; + channel_layout_ = CHANNEL_LAYOUT_NONE; + speech_type_ = kUndefined; + vad_activity_ = kVadUnknown; + profile_timestamp_ms_ = 0; + packet_infos_ = RtpPacketInfos(); + absolute_capture_timestamp_ms_ = absl::nullopt; +} + +void AudioFrame::UpdateFrame(uint32_t timestamp, + const int16_t* data, + size_t samples_per_channel, + int sample_rate_hz, + SpeechType speech_type, + VADActivity vad_activity, + size_t num_channels) { + timestamp_ = timestamp; + samples_per_channel_ = samples_per_channel; + sample_rate_hz_ = sample_rate_hz; + speech_type_ = speech_type; + vad_activity_ = vad_activity; + num_channels_ = num_channels; + channel_layout_ = GuessChannelLayout(num_channels); + if (channel_layout_ != CHANNEL_LAYOUT_UNSUPPORTED) { + RTC_DCHECK_EQ(num_channels, ChannelLayoutToChannelCount(channel_layout_)); + } + + const size_t length = samples_per_channel * num_channels; + RTC_CHECK_LE(length, kMaxDataSizeSamples); + if (data != nullptr) { + memcpy(data_, data, sizeof(int16_t) * length); + muted_ = false; + } else { + muted_ = true; + } +} + +void AudioFrame::CopyFrom(const AudioFrame& src) { + if (this == &src) + return; + + timestamp_ = src.timestamp_; + elapsed_time_ms_ = src.elapsed_time_ms_; + ntp_time_ms_ = src.ntp_time_ms_; + packet_infos_ = src.packet_infos_; + muted_ = src.muted(); + samples_per_channel_ = src.samples_per_channel_; + sample_rate_hz_ = src.sample_rate_hz_; + speech_type_ = src.speech_type_; + vad_activity_ = src.vad_activity_; + num_channels_ = src.num_channels_; + channel_layout_ = src.channel_layout_; + absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms(); + + const size_t length = samples_per_channel_ * num_channels_; + RTC_CHECK_LE(length, kMaxDataSizeSamples); + if (!src.muted()) { + memcpy(data_, src.data(), sizeof(int16_t) * length); + muted_ = false; + } +} + +void AudioFrame::UpdateProfileTimeStamp() { + profile_timestamp_ms_ = rtc::TimeMillis(); +} + +int64_t AudioFrame::ElapsedProfileTimeMs() const { + if (profile_timestamp_ms_ == 0) { + // Profiling has not been activated. + return -1; + } + return rtc::TimeSince(profile_timestamp_ms_); +} + +const int16_t* AudioFrame::data() const { + return muted_ ? empty_data() : data_; +} + +// TODO(henrik.lundin) Can we skip zeroing the buffer? +// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647. +int16_t* AudioFrame::mutable_data() { + if (muted_) { + memset(data_, 0, kMaxDataSizeBytes); + muted_ = false; + } + return data_; +} + +void AudioFrame::Mute() { + muted_ = true; +} + +bool AudioFrame::muted() const { + return muted_; +} + +// static +const int16_t* AudioFrame::empty_data() { + static int16_t* null_data = new int16_t[kMaxDataSizeSamples](); + return &null_data[0]; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/audio_frame.h b/third_party/libwebrtc/api/audio/audio_frame.h new file mode 100644 index 0000000000..d5dcb5f788 --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame.h @@ -0,0 +1,173 @@ +/* + * Copyright (c) 2018 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 API_AUDIO_AUDIO_FRAME_H_ +#define API_AUDIO_AUDIO_FRAME_H_ + +#include +#include + +#include "api/audio/channel_layout.h" +#include "api/rtp_packet_infos.h" + +namespace webrtc { + +/* This class holds up to 120 ms of super-wideband (32 kHz) stereo audio. It + * allows for adding and subtracting frames while keeping track of the resulting + * states. + * + * Notes + * - This is a de-facto api, not designed for external use. The AudioFrame class + * is in need of overhaul or even replacement, and anyone depending on it + * should be prepared for that. + * - The total number of samples is samples_per_channel_ * num_channels_. + * - Stereo data is interleaved starting with the left channel. + */ +class AudioFrame { + public: + // Using constexpr here causes linker errors unless the variable also has an + // out-of-class definition, which is impractical in this header-only class. + // (This makes no sense because it compiles as an enum value, which we most + // certainly cannot take the address of, just fine.) C++17 introduces inline + // variables which should allow us to switch to constexpr and keep this a + // header-only class. + enum : size_t { + // Stereo, 32 kHz, 120 ms (2 * 32 * 120) + // Stereo, 192 kHz, 20 ms (2 * 192 * 20) + kMaxDataSizeSamples = 7680, + kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t), + }; + + enum VADActivity { kVadActive = 0, kVadPassive = 1, kVadUnknown = 2 }; + enum SpeechType { + kNormalSpeech = 0, + kPLC = 1, + kCNG = 2, + kPLCCNG = 3, + kCodecPLC = 5, + kUndefined = 4 + }; + + AudioFrame(); + + AudioFrame(const AudioFrame&) = delete; + AudioFrame& operator=(const AudioFrame&) = delete; + + // Resets all members to their default state. + void Reset(); + // Same as Reset(), but leaves mute state unchanged. Muting a frame requires + // the buffer to be zeroed on the next call to mutable_data(). Callers + // intending to write to the buffer immediately after Reset() can instead use + // ResetWithoutMuting() to skip this wasteful zeroing. + void ResetWithoutMuting(); + + void UpdateFrame(uint32_t timestamp, + const int16_t* data, + size_t samples_per_channel, + int sample_rate_hz, + SpeechType speech_type, + VADActivity vad_activity, + size_t num_channels = 1); + + void CopyFrom(const AudioFrame& src); + + // Sets a wall-time clock timestamp in milliseconds to be used for profiling + // of time between two points in the audio chain. + // Example: + // t0: UpdateProfileTimeStamp() + // t1: ElapsedProfileTimeMs() => t1 - t0 [msec] + void UpdateProfileTimeStamp(); + // Returns the time difference between now and when UpdateProfileTimeStamp() + // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been + // called. + int64_t ElapsedProfileTimeMs() const; + + // data() returns a zeroed static buffer if the frame is muted. + // mutable_frame() always returns a non-static buffer; the first call to + // mutable_frame() zeros the non-static buffer and marks the frame unmuted. + const int16_t* data() const; + int16_t* mutable_data(); + + // Prefer to mute frames using AudioFrameOperations::Mute. + void Mute(); + // Frame is muted by default. + bool muted() const; + + size_t max_16bit_samples() const { return kMaxDataSizeSamples; } + size_t samples_per_channel() const { return samples_per_channel_; } + size_t num_channels() const { return num_channels_; } + ChannelLayout channel_layout() const { return channel_layout_; } + int sample_rate_hz() const { return sample_rate_hz_; } + + void set_absolute_capture_timestamp_ms( + int64_t absolute_capture_time_stamp_ms) { + absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms; + } + + absl::optional absolute_capture_timestamp_ms() const { + return absolute_capture_timestamp_ms_; + } + + // RTP timestamp of the first sample in the AudioFrame. + uint32_t timestamp_ = 0; + // Time since the first frame in milliseconds. + // -1 represents an uninitialized value. + int64_t elapsed_time_ms_ = -1; + // NTP time of the estimated capture time in local timebase in milliseconds. + // -1 represents an uninitialized value. + int64_t ntp_time_ms_ = -1; + size_t samples_per_channel_ = 0; + int sample_rate_hz_ = 0; + size_t num_channels_ = 0; + ChannelLayout channel_layout_ = CHANNEL_LAYOUT_NONE; + SpeechType speech_type_ = kUndefined; + VADActivity vad_activity_ = kVadUnknown; + // Monotonically increasing timestamp intended for profiling of audio frames. + // Typically used for measuring elapsed time between two different points in + // the audio path. No lock is used to save resources and we are thread safe + // by design. + // TODO(nisse@webrtc.org): consider using absl::optional. + int64_t profile_timestamp_ms_ = 0; + + // Information about packets used to assemble this audio frame. This is needed + // by `SourceTracker` when the frame is delivered to the RTCRtpReceiver's + // MediaStreamTrack, in order to implement getContributingSources(). See: + // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources + // + // TODO(bugs.webrtc.org/10757): + // Note that this information might not be fully accurate since we currently + // don't have a proper way to track it across the audio sync buffer. The + // sync buffer is the small sample-holding buffer located after the audio + // decoder and before where samples are assembled into output frames. + // + // `RtpPacketInfos` may also be empty if the audio samples did not come from + // RTP packets. E.g. if the audio were locally generated by packet loss + // concealment, comfort noise generation, etc. + RtpPacketInfos packet_infos_; + + private: + // A permanently zeroed out buffer to represent muted frames. This is a + // header-only class, so the only way to avoid creating a separate empty + // buffer per translation unit is to wrap a static in an inline function. + static const int16_t* empty_data(); + + int16_t data_[kMaxDataSizeSamples]; + bool muted_ = true; + + // Absolute capture timestamp when this audio frame was originally captured. + // This is only valid for audio frames captured on this machine. The absolute + // capture timestamp of a received frame is found in `packet_infos_`. + // This timestamp MUST be based on the same clock as rtc::TimeMillis(). + absl::optional absolute_capture_timestamp_ms_; +}; + +} // namespace webrtc + +#endif // API_AUDIO_AUDIO_FRAME_H_ diff --git a/third_party/libwebrtc/api/audio/audio_frame_api_gn/moz.build b/third_party/libwebrtc/api/audio/audio_frame_api_gn/moz.build new file mode 100644 index 0000000000..ca2c90ecfa --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame_api_gn/moz.build @@ -0,0 +1,233 @@ +# 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/api/audio/audio_frame.cc", + "/third_party/libwebrtc/api/audio/channel_layout.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("audio_frame_api_gn") diff --git a/third_party/libwebrtc/api/audio/audio_frame_processor.h b/third_party/libwebrtc/api/audio/audio_frame_processor.h new file mode 100644 index 0000000000..cb65c4817e --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame_processor.h @@ -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. + */ + +#ifndef API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ +#define API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ + +#include +#include + +namespace webrtc { + +class AudioFrame; + +// If passed into PeerConnectionFactory, will be used for additional +// processing of captured audio frames, performed before encoding. +// Implementations must be thread-safe. +class AudioFrameProcessor { + public: + using OnAudioFrameCallback = std::function)>; + virtual ~AudioFrameProcessor() = default; + + // Processes the frame received from WebRTC, is called by WebRTC off the + // realtime audio capturing path. AudioFrameProcessor must reply with + // processed frames by calling `sink_callback` if it was provided in SetSink() + // call. `sink_callback` can be called in the context of Process(). + virtual void Process(std::unique_ptr frame) = 0; + + // Atomically replaces the current sink with the new one. Before the + // first call to this function, or if the provided `sink_callback` is nullptr, + // processed frames are simply discarded. + virtual void SetSink(OnAudioFrameCallback sink_callback) = 0; +}; + +} // namespace webrtc + +#endif // API_AUDIO_AUDIO_FRAME_PROCESSOR_H_ diff --git a/third_party/libwebrtc/api/audio/audio_frame_processor_gn/moz.build b/third_party/libwebrtc/api/audio/audio_frame_processor_gn/moz.build new file mode 100644 index 0000000000..87847bb863 --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_frame_processor_gn/moz.build @@ -0,0 +1,205 @@ +# 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" +] + +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 + +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 + +if CONFIG["TARGET_CPU"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["TARGET_CPU"] == "arm": + + 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": + + 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": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64": + + DEFINES["_GNU_SOURCE"] = True + +Library("audio_frame_processor_gn") diff --git a/third_party/libwebrtc/api/audio/audio_mixer.h b/third_party/libwebrtc/api/audio/audio_mixer.h new file mode 100644 index 0000000000..3483df22bc --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_mixer.h @@ -0,0 +1,80 @@ +/* + * Copyright (c) 2011 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 API_AUDIO_AUDIO_MIXER_H_ +#define API_AUDIO_AUDIO_MIXER_H_ + +#include + +#include "api/audio/audio_frame.h" +#include "rtc_base/ref_count.h" + +namespace webrtc { + +// WORK IN PROGRESS +// This class is under development and is not yet intended for for use outside +// of WebRtc/Libjingle. +class AudioMixer : public rtc::RefCountInterface { + public: + // A callback class that all mixer participants must inherit from/implement. + class Source { + public: + enum class AudioFrameInfo { + kNormal, // The samples in audio_frame are valid and should be used. + kMuted, // The samples in audio_frame should not be used, but + // should be implicitly interpreted as zero. Other + // fields in audio_frame may be read and should + // contain meaningful values. + kError, // The audio_frame will not be used. + }; + + // Overwrites `audio_frame`. The data_ field is overwritten with + // 10 ms of new audio (either 1 or 2 interleaved channels) at + // `sample_rate_hz`. All fields in `audio_frame` must be updated. + virtual AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, + AudioFrame* audio_frame) = 0; + + // A way for a mixer implementation to distinguish participants. + virtual int Ssrc() const = 0; + + // A way for this source to say that GetAudioFrameWithInfo called + // with this sample rate or higher will not cause quality loss. + virtual int PreferredSampleRate() const = 0; + + virtual ~Source() {} + }; + + // Returns true if adding was successful. A source is never added + // twice. Addition and removal can happen on different threads. + virtual bool AddSource(Source* audio_source) = 0; + + // Removal is never attempted if a source has not been successfully + // added to the mixer. + virtual void RemoveSource(Source* audio_source) = 0; + + // Performs mixing by asking registered audio sources for audio. The + // mixed result is placed in the provided AudioFrame. This method + // will only be called from a single thread. The channels argument + // specifies the number of channels of the mix result. The mixer + // should mix at a rate that doesn't cause quality loss of the + // sources' audio. The mixing rate is one of the rates listed in + // AudioProcessing::NativeRate. All fields in + // `audio_frame_for_mixing` must be updated. + virtual void Mix(size_t number_of_channels, + AudioFrame* audio_frame_for_mixing) = 0; + + protected: + // Since the mixer is reference counted, the destructor may be + // called from any thread. + ~AudioMixer() override {} +}; +} // namespace webrtc + +#endif // API_AUDIO_AUDIO_MIXER_H_ diff --git a/third_party/libwebrtc/api/audio/audio_mixer_api_gn/moz.build b/third_party/libwebrtc/api/audio/audio_mixer_api_gn/moz.build new file mode 100644 index 0000000000..27baf1a796 --- /dev/null +++ b/third_party/libwebrtc/api/audio/audio_mixer_api_gn/moz.build @@ -0,0 +1,216 @@ +# 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" +] + +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": + + 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": + + 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": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64": + + DEFINES["_GNU_SOURCE"] = True + +Library("audio_mixer_api_gn") diff --git a/third_party/libwebrtc/api/audio/channel_layout.cc b/third_party/libwebrtc/api/audio/channel_layout.cc new file mode 100644 index 0000000000..e4ae356fab --- /dev/null +++ b/third_party/libwebrtc/api/audio/channel_layout.cc @@ -0,0 +1,282 @@ +/* + * Copyright (c) 2019 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 "api/audio/channel_layout.h" + +#include + +#include "rtc_base/arraysize.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" + +namespace webrtc { + +static const int kLayoutToChannels[] = { + 0, // CHANNEL_LAYOUT_NONE + 0, // CHANNEL_LAYOUT_UNSUPPORTED + 1, // CHANNEL_LAYOUT_MONO + 2, // CHANNEL_LAYOUT_STEREO + 3, // CHANNEL_LAYOUT_2_1 + 3, // CHANNEL_LAYOUT_SURROUND + 4, // CHANNEL_LAYOUT_4_0 + 4, // CHANNEL_LAYOUT_2_2 + 4, // CHANNEL_LAYOUT_QUAD + 5, // CHANNEL_LAYOUT_5_0 + 6, // CHANNEL_LAYOUT_5_1 + 5, // CHANNEL_LAYOUT_5_0_BACK + 6, // CHANNEL_LAYOUT_5_1_BACK + 7, // CHANNEL_LAYOUT_7_0 + 8, // CHANNEL_LAYOUT_7_1 + 8, // CHANNEL_LAYOUT_7_1_WIDE + 2, // CHANNEL_LAYOUT_STEREO_DOWNMIX + 3, // CHANNEL_LAYOUT_2POINT1 + 4, // CHANNEL_LAYOUT_3_1 + 5, // CHANNEL_LAYOUT_4_1 + 6, // CHANNEL_LAYOUT_6_0 + 6, // CHANNEL_LAYOUT_6_0_FRONT + 6, // CHANNEL_LAYOUT_HEXAGONAL + 7, // CHANNEL_LAYOUT_6_1 + 7, // CHANNEL_LAYOUT_6_1_BACK + 7, // CHANNEL_LAYOUT_6_1_FRONT + 7, // CHANNEL_LAYOUT_7_0_FRONT + 8, // CHANNEL_LAYOUT_7_1_WIDE_BACK + 8, // CHANNEL_LAYOUT_OCTAGONAL + 0, // CHANNEL_LAYOUT_DISCRETE + 3, // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC + 5, // CHANNEL_LAYOUT_4_1_QUAD_SIDE + 0, // CHANNEL_LAYOUT_BITSTREAM +}; + +// The channel orderings for each layout as specified by FFmpeg. Each value +// represents the index of each channel in each layout. Values of -1 mean the +// channel at that index is not used for that layout. For example, the left side +// surround sound channel in FFmpeg's 5.1 layout is in the 5th position (because +// the order is L, R, C, LFE, LS, RS), so +// kChannelOrderings[CHANNEL_LAYOUT_5_1][SIDE_LEFT] = 4; +static const int kChannelOrderings[CHANNEL_LAYOUT_MAX + 1][CHANNELS_MAX + 1] = { + // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR + + // CHANNEL_LAYOUT_NONE + {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_UNSUPPORTED + {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_MONO + {-1, -1, 0, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_STEREO + {0, 1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_2_1 + {0, 1, -1, -1, -1, -1, -1, -1, 2, -1, -1}, + + // CHANNEL_LAYOUT_SURROUND + {0, 1, 2, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_4_0 + {0, 1, 2, -1, -1, -1, -1, -1, 3, -1, -1}, + + // CHANNEL_LAYOUT_2_2 + {0, 1, -1, -1, -1, -1, -1, -1, -1, 2, 3}, + + // CHANNEL_LAYOUT_QUAD + {0, 1, -1, -1, 2, 3, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_5_0 + {0, 1, 2, -1, -1, -1, -1, -1, -1, 3, 4}, + + // CHANNEL_LAYOUT_5_1 + {0, 1, 2, 3, -1, -1, -1, -1, -1, 4, 5}, + + // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR + + // CHANNEL_LAYOUT_5_0_BACK + {0, 1, 2, -1, 3, 4, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_5_1_BACK + {0, 1, 2, 3, 4, 5, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_7_0 + {0, 1, 2, -1, 5, 6, -1, -1, -1, 3, 4}, + + // CHANNEL_LAYOUT_7_1 + {0, 1, 2, 3, 6, 7, -1, -1, -1, 4, 5}, + + // CHANNEL_LAYOUT_7_1_WIDE + {0, 1, 2, 3, -1, -1, 6, 7, -1, 4, 5}, + + // CHANNEL_LAYOUT_STEREO_DOWNMIX + {0, 1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_2POINT1 + {0, 1, -1, 2, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_3_1 + {0, 1, 2, 3, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_4_1 + {0, 1, 2, 4, -1, -1, -1, -1, 3, -1, -1}, + + // CHANNEL_LAYOUT_6_0 + {0, 1, 2, -1, -1, -1, -1, -1, 5, 3, 4}, + + // CHANNEL_LAYOUT_6_0_FRONT + {0, 1, -1, -1, -1, -1, 4, 5, -1, 2, 3}, + + // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR + + // CHANNEL_LAYOUT_HEXAGONAL + {0, 1, 2, -1, 3, 4, -1, -1, 5, -1, -1}, + + // CHANNEL_LAYOUT_6_1 + {0, 1, 2, 3, -1, -1, -1, -1, 6, 4, 5}, + + // CHANNEL_LAYOUT_6_1_BACK + {0, 1, 2, 3, 4, 5, -1, -1, 6, -1, -1}, + + // CHANNEL_LAYOUT_6_1_FRONT + {0, 1, -1, 6, -1, -1, 4, 5, -1, 2, 3}, + + // CHANNEL_LAYOUT_7_0_FRONT + {0, 1, 2, -1, -1, -1, 5, 6, -1, 3, 4}, + + // CHANNEL_LAYOUT_7_1_WIDE_BACK + {0, 1, 2, 3, 4, 5, 6, 7, -1, -1, -1}, + + // CHANNEL_LAYOUT_OCTAGONAL + {0, 1, 2, -1, 5, 6, -1, -1, 7, 3, 4}, + + // CHANNEL_LAYOUT_DISCRETE + {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC + {0, 1, 2, -1, -1, -1, -1, -1, -1, -1, -1}, + + // CHANNEL_LAYOUT_4_1_QUAD_SIDE + {0, 1, -1, 4, -1, -1, -1, -1, -1, 2, 3}, + + // CHANNEL_LAYOUT_BITSTREAM + {-1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1}, + + // FL | FR | FC | LFE | BL | BR | FLofC | FRofC | BC | SL | SR +}; + +int ChannelLayoutToChannelCount(ChannelLayout layout) { + RTC_DCHECK_LT(static_cast(layout), arraysize(kLayoutToChannels)); + RTC_DCHECK_LE(kLayoutToChannels[layout], kMaxConcurrentChannels); + return kLayoutToChannels[layout]; +} + +// Converts a channel count into a channel layout. +ChannelLayout GuessChannelLayout(int channels) { + switch (channels) { + case 1: + return CHANNEL_LAYOUT_MONO; + case 2: + return CHANNEL_LAYOUT_STEREO; + case 3: + return CHANNEL_LAYOUT_SURROUND; + case 4: + return CHANNEL_LAYOUT_QUAD; + case 5: + return CHANNEL_LAYOUT_5_0; + case 6: + return CHANNEL_LAYOUT_5_1; + case 7: + return CHANNEL_LAYOUT_6_1; + case 8: + return CHANNEL_LAYOUT_7_1; + default: + RTC_DLOG(LS_WARNING) << "Unsupported channel count: " << channels; + } + return CHANNEL_LAYOUT_UNSUPPORTED; +} + +int ChannelOrder(ChannelLayout layout, Channels channel) { + RTC_DCHECK_LT(static_cast(layout), arraysize(kChannelOrderings)); + RTC_DCHECK_LT(static_cast(channel), arraysize(kChannelOrderings[0])); + return kChannelOrderings[layout][channel]; +} + +const char* ChannelLayoutToString(ChannelLayout layout) { + switch (layout) { + case CHANNEL_LAYOUT_NONE: + return "NONE"; + case CHANNEL_LAYOUT_UNSUPPORTED: + return "UNSUPPORTED"; + case CHANNEL_LAYOUT_MONO: + return "MONO"; + case CHANNEL_LAYOUT_STEREO: + return "STEREO"; + case CHANNEL_LAYOUT_2_1: + return "2.1"; + case CHANNEL_LAYOUT_SURROUND: + return "SURROUND"; + case CHANNEL_LAYOUT_4_0: + return "4.0"; + case CHANNEL_LAYOUT_2_2: + return "QUAD_SIDE"; + case CHANNEL_LAYOUT_QUAD: + return "QUAD"; + case CHANNEL_LAYOUT_5_0: + return "5.0"; + case CHANNEL_LAYOUT_5_1: + return "5.1"; + case CHANNEL_LAYOUT_5_0_BACK: + return "5.0_BACK"; + case CHANNEL_LAYOUT_5_1_BACK: + return "5.1_BACK"; + case CHANNEL_LAYOUT_7_0: + return "7.0"; + case CHANNEL_LAYOUT_7_1: + return "7.1"; + case CHANNEL_LAYOUT_7_1_WIDE: + return "7.1_WIDE"; + case CHANNEL_LAYOUT_STEREO_DOWNMIX: + return "STEREO_DOWNMIX"; + case CHANNEL_LAYOUT_2POINT1: + return "2POINT1"; + case CHANNEL_LAYOUT_3_1: + return "3.1"; + case CHANNEL_LAYOUT_4_1: + return "4.1"; + case CHANNEL_LAYOUT_6_0: + return "6.0"; + case CHANNEL_LAYOUT_6_0_FRONT: + return "6.0_FRONT"; + case CHANNEL_LAYOUT_HEXAGONAL: + return "HEXAGONAL"; + case CHANNEL_LAYOUT_6_1: + return "6.1"; + case CHANNEL_LAYOUT_6_1_BACK: + return "6.1_BACK"; + case CHANNEL_LAYOUT_6_1_FRONT: + return "6.1_FRONT"; + case CHANNEL_LAYOUT_7_0_FRONT: + return "7.0_FRONT"; + case CHANNEL_LAYOUT_7_1_WIDE_BACK: + return "7.1_WIDE_BACK"; + case CHANNEL_LAYOUT_OCTAGONAL: + return "OCTAGONAL"; + case CHANNEL_LAYOUT_DISCRETE: + return "DISCRETE"; + case CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC: + return "STEREO_AND_KEYBOARD_MIC"; + case CHANNEL_LAYOUT_4_1_QUAD_SIDE: + return "4.1_QUAD_SIDE"; + case CHANNEL_LAYOUT_BITSTREAM: + return "BITSTREAM"; + } + RTC_DCHECK_NOTREACHED() << "Invalid channel layout provided: " << layout; + return ""; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/channel_layout.h b/third_party/libwebrtc/api/audio/channel_layout.h new file mode 100644 index 0000000000..175aee71e5 --- /dev/null +++ b/third_party/libwebrtc/api/audio/channel_layout.h @@ -0,0 +1,165 @@ +/* + * Copyright (c) 2019 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 API_AUDIO_CHANNEL_LAYOUT_H_ +#define API_AUDIO_CHANNEL_LAYOUT_H_ + +namespace webrtc { + +// This file is derived from Chromium's base/channel_layout.h. + +// Enumerates the various representations of the ordering of audio channels. +// Logged to UMA, so never reuse a value, always add new/greater ones! +enum ChannelLayout { + CHANNEL_LAYOUT_NONE = 0, + CHANNEL_LAYOUT_UNSUPPORTED = 1, + + // Front C + CHANNEL_LAYOUT_MONO = 2, + + // Front L, Front R + CHANNEL_LAYOUT_STEREO = 3, + + // Front L, Front R, Back C + CHANNEL_LAYOUT_2_1 = 4, + + // Front L, Front R, Front C + CHANNEL_LAYOUT_SURROUND = 5, + + // Front L, Front R, Front C, Back C + CHANNEL_LAYOUT_4_0 = 6, + + // Front L, Front R, Side L, Side R + CHANNEL_LAYOUT_2_2 = 7, + + // Front L, Front R, Back L, Back R + CHANNEL_LAYOUT_QUAD = 8, + + // Front L, Front R, Front C, Side L, Side R + CHANNEL_LAYOUT_5_0 = 9, + + // Front L, Front R, Front C, LFE, Side L, Side R + CHANNEL_LAYOUT_5_1 = 10, + + // Front L, Front R, Front C, Back L, Back R + CHANNEL_LAYOUT_5_0_BACK = 11, + + // Front L, Front R, Front C, LFE, Back L, Back R + CHANNEL_LAYOUT_5_1_BACK = 12, + + // Front L, Front R, Front C, Side L, Side R, Back L, Back R + CHANNEL_LAYOUT_7_0 = 13, + + // Front L, Front R, Front C, LFE, Side L, Side R, Back L, Back R + CHANNEL_LAYOUT_7_1 = 14, + + // Front L, Front R, Front C, LFE, Side L, Side R, Front LofC, Front RofC + CHANNEL_LAYOUT_7_1_WIDE = 15, + + // Stereo L, Stereo R + CHANNEL_LAYOUT_STEREO_DOWNMIX = 16, + + // Stereo L, Stereo R, LFE + CHANNEL_LAYOUT_2POINT1 = 17, + + // Stereo L, Stereo R, Front C, LFE + CHANNEL_LAYOUT_3_1 = 18, + + // Stereo L, Stereo R, Front C, Rear C, LFE + CHANNEL_LAYOUT_4_1 = 19, + + // Stereo L, Stereo R, Front C, Side L, Side R, Back C + CHANNEL_LAYOUT_6_0 = 20, + + // Stereo L, Stereo R, Side L, Side R, Front LofC, Front RofC + CHANNEL_LAYOUT_6_0_FRONT = 21, + + // Stereo L, Stereo R, Front C, Rear L, Rear R, Rear C + CHANNEL_LAYOUT_HEXAGONAL = 22, + + // Stereo L, Stereo R, Front C, LFE, Side L, Side R, Rear Center + CHANNEL_LAYOUT_6_1 = 23, + + // Stereo L, Stereo R, Front C, LFE, Back L, Back R, Rear Center + CHANNEL_LAYOUT_6_1_BACK = 24, + + // Stereo L, Stereo R, Side L, Side R, Front LofC, Front RofC, LFE + CHANNEL_LAYOUT_6_1_FRONT = 25, + + // Front L, Front R, Front C, Side L, Side R, Front LofC, Front RofC + CHANNEL_LAYOUT_7_0_FRONT = 26, + + // Front L, Front R, Front C, LFE, Back L, Back R, Front LofC, Front RofC + CHANNEL_LAYOUT_7_1_WIDE_BACK = 27, + + // Front L, Front R, Front C, Side L, Side R, Rear L, Back R, Back C. + CHANNEL_LAYOUT_OCTAGONAL = 28, + + // Channels are not explicitly mapped to speakers. + CHANNEL_LAYOUT_DISCRETE = 29, + + // Front L, Front R, Front C. Front C contains the keyboard mic audio. This + // layout is only intended for input for WebRTC. The Front C channel + // is stripped away in the WebRTC audio input pipeline and never seen outside + // of that. + CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC = 30, + + // Front L, Front R, Side L, Side R, LFE + CHANNEL_LAYOUT_4_1_QUAD_SIDE = 31, + + // Actual channel layout is specified in the bitstream and the actual channel + // count is unknown at Chromium media pipeline level (useful for audio + // pass-through mode). + CHANNEL_LAYOUT_BITSTREAM = 32, + + // Max value, must always equal the largest entry ever logged. + CHANNEL_LAYOUT_MAX = CHANNEL_LAYOUT_BITSTREAM +}; + +// Note: Do not reorder or reassign these values; other code depends on their +// ordering to operate correctly. E.g., CoreAudio channel layout computations. +enum Channels { + LEFT = 0, + RIGHT, + CENTER, + LFE, + BACK_LEFT, + BACK_RIGHT, + LEFT_OF_CENTER, + RIGHT_OF_CENTER, + BACK_CENTER, + SIDE_LEFT, + SIDE_RIGHT, + CHANNELS_MAX = + SIDE_RIGHT, // Must always equal the largest value ever logged. +}; + +// The maximum number of concurrently active channels for all possible layouts. +// ChannelLayoutToChannelCount() will never return a value higher than this. +constexpr int kMaxConcurrentChannels = 8; + +// Returns the expected channel position in an interleaved stream. Values of -1 +// mean the channel at that index is not used for that layout. Values range +// from 0 to ChannelLayoutToChannelCount(layout) - 1. +int ChannelOrder(ChannelLayout layout, Channels channel); + +// Returns the number of channels in a given ChannelLayout. +int ChannelLayoutToChannelCount(ChannelLayout layout); + +// Given the number of channels, return the best layout, +// or return CHANNEL_LAYOUT_UNSUPPORTED if there is no good match. +ChannelLayout GuessChannelLayout(int channels); + +// Returns a string representation of the channel layout. +const char* ChannelLayoutToString(ChannelLayout layout); + +} // namespace webrtc + +#endif // API_AUDIO_CHANNEL_LAYOUT_H_ diff --git a/third_party/libwebrtc/api/audio/echo_canceller3_config.cc b/third_party/libwebrtc/api/audio/echo_canceller3_config.cc new file mode 100644 index 0000000000..0224c712b4 --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_canceller3_config.cc @@ -0,0 +1,278 @@ +/* + * Copyright (c) 2018 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 "api/audio/echo_canceller3_config.h" + +#include +#include + +#include "rtc_base/checks.h" +#include "rtc_base/numerics/safe_minmax.h" + +namespace webrtc { +namespace { +bool Limit(float* value, float min, float max) { + float clamped = rtc::SafeClamp(*value, min, max); + clamped = std::isfinite(clamped) ? clamped : min; + bool res = *value == clamped; + *value = clamped; + return res; +} + +bool Limit(size_t* value, size_t min, size_t max) { + size_t clamped = rtc::SafeClamp(*value, min, max); + bool res = *value == clamped; + *value = clamped; + return res; +} + +bool Limit(int* value, int min, int max) { + int clamped = rtc::SafeClamp(*value, min, max); + bool res = *value == clamped; + *value = clamped; + return res; +} + +bool FloorLimit(size_t* value, size_t min) { + size_t clamped = *value >= min ? *value : min; + bool res = *value == clamped; + *value = clamped; + return res; +} + +} // namespace + +EchoCanceller3Config::EchoCanceller3Config() = default; +EchoCanceller3Config::EchoCanceller3Config(const EchoCanceller3Config& e) = + default; +EchoCanceller3Config& EchoCanceller3Config::operator=( + const EchoCanceller3Config& e) = default; +EchoCanceller3Config::Delay::Delay() = default; +EchoCanceller3Config::Delay::Delay(const EchoCanceller3Config::Delay& e) = + default; +EchoCanceller3Config::Delay& EchoCanceller3Config::Delay::operator=( + const Delay& e) = default; + +EchoCanceller3Config::EchoModel::EchoModel() = default; +EchoCanceller3Config::EchoModel::EchoModel( + const EchoCanceller3Config::EchoModel& e) = default; +EchoCanceller3Config::EchoModel& EchoCanceller3Config::EchoModel::operator=( + const EchoModel& e) = default; + +EchoCanceller3Config::Suppressor::Suppressor() = default; +EchoCanceller3Config::Suppressor::Suppressor( + const EchoCanceller3Config::Suppressor& e) = default; +EchoCanceller3Config::Suppressor& EchoCanceller3Config::Suppressor::operator=( + const Suppressor& e) = default; + +EchoCanceller3Config::Suppressor::MaskingThresholds::MaskingThresholds( + float enr_transparent, + float enr_suppress, + float emr_transparent) + : enr_transparent(enr_transparent), + enr_suppress(enr_suppress), + emr_transparent(emr_transparent) {} +EchoCanceller3Config::Suppressor::MaskingThresholds::MaskingThresholds( + const EchoCanceller3Config::Suppressor::MaskingThresholds& e) = default; +EchoCanceller3Config::Suppressor::MaskingThresholds& +EchoCanceller3Config::Suppressor::MaskingThresholds::operator=( + const MaskingThresholds& e) = default; + +EchoCanceller3Config::Suppressor::Tuning::Tuning(MaskingThresholds mask_lf, + MaskingThresholds mask_hf, + float max_inc_factor, + float max_dec_factor_lf) + : mask_lf(mask_lf), + mask_hf(mask_hf), + max_inc_factor(max_inc_factor), + max_dec_factor_lf(max_dec_factor_lf) {} +EchoCanceller3Config::Suppressor::Tuning::Tuning( + const EchoCanceller3Config::Suppressor::Tuning& e) = default; +EchoCanceller3Config::Suppressor::Tuning& +EchoCanceller3Config::Suppressor::Tuning::operator=(const Tuning& e) = default; + +bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) { + RTC_DCHECK(config); + EchoCanceller3Config* c = config; + bool res = true; + + if (c->delay.down_sampling_factor != 4 && + c->delay.down_sampling_factor != 8) { + c->delay.down_sampling_factor = 4; + res = false; + } + + res = res & Limit(&c->delay.default_delay, 0, 5000); + res = res & Limit(&c->delay.num_filters, 0, 5000); + res = res & Limit(&c->delay.delay_headroom_samples, 0, 5000); + res = res & Limit(&c->delay.hysteresis_limit_blocks, 0, 5000); + res = res & Limit(&c->delay.fixed_capture_delay_samples, 0, 5000); + res = res & Limit(&c->delay.delay_estimate_smoothing, 0.f, 1.f); + res = res & Limit(&c->delay.delay_candidate_detection_threshold, 0.f, 1.f); + res = res & Limit(&c->delay.delay_selection_thresholds.initial, 1, 250); + res = res & Limit(&c->delay.delay_selection_thresholds.converged, 1, 250); + + res = res & FloorLimit(&c->filter.refined.length_blocks, 1); + res = res & Limit(&c->filter.refined.leakage_converged, 0.f, 1000.f); + res = res & Limit(&c->filter.refined.leakage_diverged, 0.f, 1000.f); + res = res & Limit(&c->filter.refined.error_floor, 0.f, 1000.f); + res = res & Limit(&c->filter.refined.error_ceil, 0.f, 100000000.f); + res = res & Limit(&c->filter.refined.noise_gate, 0.f, 100000000.f); + + res = res & FloorLimit(&c->filter.refined_initial.length_blocks, 1); + res = res & Limit(&c->filter.refined_initial.leakage_converged, 0.f, 1000.f); + res = res & Limit(&c->filter.refined_initial.leakage_diverged, 0.f, 1000.f); + res = res & Limit(&c->filter.refined_initial.error_floor, 0.f, 1000.f); + res = res & Limit(&c->filter.refined_initial.error_ceil, 0.f, 100000000.f); + res = res & Limit(&c->filter.refined_initial.noise_gate, 0.f, 100000000.f); + + if (c->filter.refined.length_blocks < + c->filter.refined_initial.length_blocks) { + c->filter.refined_initial.length_blocks = c->filter.refined.length_blocks; + res = false; + } + + res = res & FloorLimit(&c->filter.coarse.length_blocks, 1); + res = res & Limit(&c->filter.coarse.rate, 0.f, 1.f); + res = res & Limit(&c->filter.coarse.noise_gate, 0.f, 100000000.f); + + res = res & FloorLimit(&c->filter.coarse_initial.length_blocks, 1); + res = res & Limit(&c->filter.coarse_initial.rate, 0.f, 1.f); + res = res & Limit(&c->filter.coarse_initial.noise_gate, 0.f, 100000000.f); + + if (c->filter.coarse.length_blocks < c->filter.coarse_initial.length_blocks) { + c->filter.coarse_initial.length_blocks = c->filter.coarse.length_blocks; + res = false; + } + + res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000); + res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f); + res = res & Limit(&c->filter.coarse_reset_hangover_blocks, 0, 250000); + + res = res & Limit(&c->erle.min, 1.f, 100000.f); + res = res & Limit(&c->erle.max_l, 1.f, 100000.f); + res = res & Limit(&c->erle.max_h, 1.f, 100000.f); + if (c->erle.min > c->erle.max_l || c->erle.min > c->erle.max_h) { + c->erle.min = std::min(c->erle.max_l, c->erle.max_h); + res = false; + } + res = res & Limit(&c->erle.num_sections, 1, c->filter.refined.length_blocks); + + res = res & Limit(&c->ep_strength.default_gain, 0.f, 1000000.f); + res = res & Limit(&c->ep_strength.default_len, -1.f, 1.f); + res = res & Limit(&c->ep_strength.nearend_len, -1.0f, 1.0f); + + res = + res & Limit(&c->echo_audibility.low_render_limit, 0.f, 32768.f * 32768.f); + res = res & + Limit(&c->echo_audibility.normal_render_limit, 0.f, 32768.f * 32768.f); + res = res & Limit(&c->echo_audibility.floor_power, 0.f, 32768.f * 32768.f); + res = res & Limit(&c->echo_audibility.audibility_threshold_lf, 0.f, + 32768.f * 32768.f); + res = res & Limit(&c->echo_audibility.audibility_threshold_mf, 0.f, + 32768.f * 32768.f); + res = res & Limit(&c->echo_audibility.audibility_threshold_hf, 0.f, + 32768.f * 32768.f); + + res = res & + Limit(&c->render_levels.active_render_limit, 0.f, 32768.f * 32768.f); + res = res & Limit(&c->render_levels.poor_excitation_render_limit, 0.f, + 32768.f * 32768.f); + res = res & Limit(&c->render_levels.poor_excitation_render_limit_ds8, 0.f, + 32768.f * 32768.f); + + res = res & Limit(&c->echo_model.noise_floor_hold, 0, 1000); + res = res & Limit(&c->echo_model.min_noise_floor_power, 0, 2000000.f); + res = res & Limit(&c->echo_model.stationary_gate_slope, 0, 1000000.f); + res = res & Limit(&c->echo_model.noise_gate_power, 0, 1000000.f); + res = res & Limit(&c->echo_model.noise_gate_slope, 0, 1000000.f); + res = res & Limit(&c->echo_model.render_pre_window_size, 0, 100); + res = res & Limit(&c->echo_model.render_post_window_size, 0, 100); + + res = res & Limit(&c->comfort_noise.noise_floor_dbfs, -200.f, 0.f); + + res = res & Limit(&c->suppressor.nearend_average_blocks, 1, 5000); + + res = res & + Limit(&c->suppressor.normal_tuning.mask_lf.enr_transparent, 0.f, 100.f); + res = res & + Limit(&c->suppressor.normal_tuning.mask_lf.enr_suppress, 0.f, 100.f); + res = res & + Limit(&c->suppressor.normal_tuning.mask_lf.emr_transparent, 0.f, 100.f); + res = res & + Limit(&c->suppressor.normal_tuning.mask_hf.enr_transparent, 0.f, 100.f); + res = res & + Limit(&c->suppressor.normal_tuning.mask_hf.enr_suppress, 0.f, 100.f); + res = res & + Limit(&c->suppressor.normal_tuning.mask_hf.emr_transparent, 0.f, 100.f); + res = res & Limit(&c->suppressor.normal_tuning.max_inc_factor, 0.f, 100.f); + res = res & Limit(&c->suppressor.normal_tuning.max_dec_factor_lf, 0.f, 100.f); + + res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.enr_transparent, 0.f, + 100.f); + res = res & + Limit(&c->suppressor.nearend_tuning.mask_lf.enr_suppress, 0.f, 100.f); + res = res & Limit(&c->suppressor.nearend_tuning.mask_lf.emr_transparent, 0.f, + 100.f); + res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.enr_transparent, 0.f, + 100.f); + res = res & + Limit(&c->suppressor.nearend_tuning.mask_hf.enr_suppress, 0.f, 100.f); + res = res & Limit(&c->suppressor.nearend_tuning.mask_hf.emr_transparent, 0.f, + 100.f); + res = res & Limit(&c->suppressor.nearend_tuning.max_inc_factor, 0.f, 100.f); + res = + res & Limit(&c->suppressor.nearend_tuning.max_dec_factor_lf, 0.f, 100.f); + + res = res & Limit(&c->suppressor.last_permanent_lf_smoothing_band, 0, 64); + res = res & Limit(&c->suppressor.last_lf_smoothing_band, 0, 64); + res = res & Limit(&c->suppressor.last_lf_band, 0, 63); + res = res & + Limit(&c->suppressor.first_hf_band, c->suppressor.last_lf_band + 1, 64); + + res = res & Limit(&c->suppressor.dominant_nearend_detection.enr_threshold, + 0.f, 1000000.f); + res = res & Limit(&c->suppressor.dominant_nearend_detection.snr_threshold, + 0.f, 1000000.f); + res = res & Limit(&c->suppressor.dominant_nearend_detection.hold_duration, 0, + 10000); + res = res & Limit(&c->suppressor.dominant_nearend_detection.trigger_threshold, + 0, 10000); + + res = res & + Limit(&c->suppressor.subband_nearend_detection.nearend_average_blocks, + 1, 1024); + res = + res & Limit(&c->suppressor.subband_nearend_detection.subband1.low, 0, 65); + res = res & Limit(&c->suppressor.subband_nearend_detection.subband1.high, + c->suppressor.subband_nearend_detection.subband1.low, 65); + res = + res & Limit(&c->suppressor.subband_nearend_detection.subband2.low, 0, 65); + res = res & Limit(&c->suppressor.subband_nearend_detection.subband2.high, + c->suppressor.subband_nearend_detection.subband2.low, 65); + res = res & Limit(&c->suppressor.subband_nearend_detection.nearend_threshold, + 0.f, 1.e24f); + res = res & Limit(&c->suppressor.subband_nearend_detection.snr_threshold, 0.f, + 1.e24f); + + res = res & Limit(&c->suppressor.high_bands_suppression.enr_threshold, 0.f, + 1000000.f); + res = res & Limit(&c->suppressor.high_bands_suppression.max_gain_during_echo, + 0.f, 1.f); + res = res & Limit(&c->suppressor.high_bands_suppression + .anti_howling_activation_threshold, + 0.f, 32768.f * 32768.f); + res = res & Limit(&c->suppressor.high_bands_suppression.anti_howling_gain, + 0.f, 1.f); + + res = res & Limit(&c->suppressor.floor_first_increase, 0.f, 1000000.f); + + return res; +} +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/echo_canceller3_config.h b/third_party/libwebrtc/api/audio/echo_canceller3_config.h new file mode 100644 index 0000000000..4b1c7fbc47 --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_canceller3_config.h @@ -0,0 +1,250 @@ +/* + * Copyright (c) 2018 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 API_AUDIO_ECHO_CANCELLER3_CONFIG_H_ +#define API_AUDIO_ECHO_CANCELLER3_CONFIG_H_ + +#include // size_t + +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Configuration struct for EchoCanceller3 +struct RTC_EXPORT EchoCanceller3Config { + // Checks and updates the config parameters to lie within (mostly) reasonable + // ranges. Returns true if and only of the config did not need to be changed. + static bool Validate(EchoCanceller3Config* config); + + EchoCanceller3Config(); + EchoCanceller3Config(const EchoCanceller3Config& e); + EchoCanceller3Config& operator=(const EchoCanceller3Config& other); + + struct Buffering { + size_t excess_render_detection_interval_blocks = 250; + size_t max_allowed_excess_render_blocks = 8; + } buffering; + + struct Delay { + Delay(); + Delay(const Delay& e); + Delay& operator=(const Delay& e); + size_t default_delay = 5; + size_t down_sampling_factor = 4; + size_t num_filters = 5; + size_t delay_headroom_samples = 32; + size_t hysteresis_limit_blocks = 1; + size_t fixed_capture_delay_samples = 0; + float delay_estimate_smoothing = 0.7f; + float delay_estimate_smoothing_delay_found = 0.7f; + float delay_candidate_detection_threshold = 0.2f; + struct DelaySelectionThresholds { + int initial; + int converged; + } delay_selection_thresholds = {5, 20}; + bool use_external_delay_estimator = false; + bool log_warning_on_delay_changes = false; + struct AlignmentMixing { + bool downmix; + bool adaptive_selection; + float activity_power_threshold; + bool prefer_first_two_channels; + }; + AlignmentMixing render_alignment_mixing = {false, true, 10000.f, true}; + AlignmentMixing capture_alignment_mixing = {false, true, 10000.f, false}; + bool detect_pre_echo = true; + } delay; + + struct Filter { + struct RefinedConfiguration { + size_t length_blocks; + float leakage_converged; + float leakage_diverged; + float error_floor; + float error_ceil; + float noise_gate; + }; + + struct CoarseConfiguration { + size_t length_blocks; + float rate; + float noise_gate; + }; + + RefinedConfiguration refined = {13, 0.00005f, 0.05f, + 0.001f, 2.f, 20075344.f}; + CoarseConfiguration coarse = {13, 0.7f, 20075344.f}; + + RefinedConfiguration refined_initial = {12, 0.005f, 0.5f, + 0.001f, 2.f, 20075344.f}; + CoarseConfiguration coarse_initial = {12, 0.9f, 20075344.f}; + + size_t config_change_duration_blocks = 250; + float initial_state_seconds = 2.5f; + int coarse_reset_hangover_blocks = 25; + bool conservative_initial_phase = false; + bool enable_coarse_filter_output_usage = true; + bool use_linear_filter = true; + bool high_pass_filter_echo_reference = false; + bool export_linear_aec_output = false; + } filter; + + struct Erle { + float min = 1.f; + float max_l = 4.f; + float max_h = 1.5f; + bool onset_detection = true; + size_t num_sections = 1; + bool clamp_quality_estimate_to_zero = true; + bool clamp_quality_estimate_to_one = true; + } erle; + + struct EpStrength { + float default_gain = 1.f; + float default_len = 0.83f; + float nearend_len = 0.83f; + bool echo_can_saturate = true; + bool bounded_erl = false; + bool erle_onset_compensation_in_dominant_nearend = false; + bool use_conservative_tail_frequency_response = true; + } ep_strength; + + struct EchoAudibility { + float low_render_limit = 4 * 64.f; + float normal_render_limit = 64.f; + float floor_power = 2 * 64.f; + float audibility_threshold_lf = 10; + float audibility_threshold_mf = 10; + float audibility_threshold_hf = 10; + bool use_stationarity_properties = false; + bool use_stationarity_properties_at_init = false; + } echo_audibility; + + struct RenderLevels { + float active_render_limit = 100.f; + float poor_excitation_render_limit = 150.f; + float poor_excitation_render_limit_ds8 = 20.f; + float render_power_gain_db = 0.f; + } render_levels; + + struct EchoRemovalControl { + bool has_clock_drift = false; + bool linear_and_stable_echo_path = false; + } echo_removal_control; + + struct EchoModel { + EchoModel(); + EchoModel(const EchoModel& e); + EchoModel& operator=(const EchoModel& e); + size_t noise_floor_hold = 50; + float min_noise_floor_power = 1638400.f; + float stationary_gate_slope = 10.f; + float noise_gate_power = 27509.42f; + float noise_gate_slope = 0.3f; + size_t render_pre_window_size = 1; + size_t render_post_window_size = 1; + bool model_reverb_in_nonlinear_mode = true; + } echo_model; + + struct ComfortNoise { + float noise_floor_dbfs = -96.03406f; + } comfort_noise; + + struct Suppressor { + Suppressor(); + Suppressor(const Suppressor& e); + Suppressor& operator=(const Suppressor& e); + + size_t nearend_average_blocks = 4; + + struct MaskingThresholds { + MaskingThresholds(float enr_transparent, + float enr_suppress, + float emr_transparent); + MaskingThresholds(const MaskingThresholds& e); + MaskingThresholds& operator=(const MaskingThresholds& e); + float enr_transparent; + float enr_suppress; + float emr_transparent; + }; + + struct Tuning { + Tuning(MaskingThresholds mask_lf, + MaskingThresholds mask_hf, + float max_inc_factor, + float max_dec_factor_lf); + Tuning(const Tuning& e); + Tuning& operator=(const Tuning& e); + MaskingThresholds mask_lf; + MaskingThresholds mask_hf; + float max_inc_factor; + float max_dec_factor_lf; + }; + + Tuning normal_tuning = Tuning(MaskingThresholds(.3f, .4f, .3f), + MaskingThresholds(.07f, .1f, .3f), + 2.0f, + 0.25f); + Tuning nearend_tuning = Tuning(MaskingThresholds(1.09f, 1.1f, .3f), + MaskingThresholds(.1f, .3f, .3f), + 2.0f, + 0.25f); + + bool lf_smoothing_during_initial_phase = true; + int last_permanent_lf_smoothing_band = 0; + int last_lf_smoothing_band = 5; + int last_lf_band = 5; + int first_hf_band = 8; + + struct DominantNearendDetection { + float enr_threshold = .25f; + float enr_exit_threshold = 10.f; + float snr_threshold = 30.f; + int hold_duration = 50; + int trigger_threshold = 12; + bool use_during_initial_phase = true; + bool use_unbounded_echo_spectrum = true; + } dominant_nearend_detection; + + struct SubbandNearendDetection { + size_t nearend_average_blocks = 1; + struct SubbandRegion { + size_t low; + size_t high; + }; + SubbandRegion subband1 = {1, 1}; + SubbandRegion subband2 = {1, 1}; + float nearend_threshold = 1.f; + float snr_threshold = 1.f; + } subband_nearend_detection; + + bool use_subband_nearend_detection = false; + + struct HighBandsSuppression { + float enr_threshold = 1.f; + float max_gain_during_echo = 1.f; + float anti_howling_activation_threshold = 400.f; + float anti_howling_gain = 1.f; + } high_bands_suppression; + + float floor_first_increase = 0.00001f; + bool conservative_hf_suppression = false; + } suppressor; + + struct MultiChannel { + bool detect_stereo_content = true; + float stereo_detection_threshold = 0.0f; + int stereo_detection_timeout_threshold_seconds = 300; + float stereo_detection_hysteresis_seconds = 2.0f; + } multi_channel; +}; +} // namespace webrtc + +#endif // API_AUDIO_ECHO_CANCELLER3_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio/echo_canceller3_factory.cc b/third_party/libwebrtc/api/audio/echo_canceller3_factory.cc new file mode 100644 index 0000000000..284b117bea --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_canceller3_factory.cc @@ -0,0 +1,32 @@ +/* + * Copyright (c) 2018 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 "api/audio/echo_canceller3_factory.h" + +#include + +#include "modules/audio_processing/aec3/echo_canceller3.h" + +namespace webrtc { + +EchoCanceller3Factory::EchoCanceller3Factory() {} + +EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config& config) + : config_(config) {} + +std::unique_ptr EchoCanceller3Factory::Create( + int sample_rate_hz, + int num_render_channels, + int num_capture_channels) { + return std::make_unique( + config_, /*multichannel_config=*/absl::nullopt, sample_rate_hz, + num_render_channels, num_capture_channels); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/echo_canceller3_factory.h b/third_party/libwebrtc/api/audio/echo_canceller3_factory.h new file mode 100644 index 0000000000..8b5380057b --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_canceller3_factory.h @@ -0,0 +1,41 @@ +/* + * Copyright (c) 2018 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 API_AUDIO_ECHO_CANCELLER3_FACTORY_H_ +#define API_AUDIO_ECHO_CANCELLER3_FACTORY_H_ + +#include + +#include "api/audio/echo_canceller3_config.h" +#include "api/audio/echo_control.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +class RTC_EXPORT EchoCanceller3Factory : public EchoControlFactory { + public: + // Factory producing EchoCanceller3 instances with the default configuration. + EchoCanceller3Factory(); + + // Factory producing EchoCanceller3 instances with the specified + // configuration. + explicit EchoCanceller3Factory(const EchoCanceller3Config& config); + + // Creates an EchoCanceller3 with a specified channel count and sampling rate. + std::unique_ptr Create(int sample_rate_hz, + int num_render_channels, + int num_capture_channels) override; + + private: + const EchoCanceller3Config config_; +}; +} // namespace webrtc + +#endif // API_AUDIO_ECHO_CANCELLER3_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio/echo_control.h b/third_party/libwebrtc/api/audio/echo_control.h new file mode 100644 index 0000000000..74fbc27b12 --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_control.h @@ -0,0 +1,75 @@ +/* + * Copyright (c) 2018 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 API_AUDIO_ECHO_CONTROL_H_ +#define API_AUDIO_ECHO_CONTROL_H_ + +#include + +#include "rtc_base/checks.h" + +namespace webrtc { + +class AudioBuffer; + +// Interface for an acoustic echo cancellation (AEC) submodule. +class EchoControl { + public: + // Analysis (not changing) of the render signal. + virtual void AnalyzeRender(AudioBuffer* render) = 0; + + // Analysis (not changing) of the capture signal. + virtual void AnalyzeCapture(AudioBuffer* capture) = 0; + + // Processes the capture signal in order to remove the echo. + virtual void ProcessCapture(AudioBuffer* capture, bool level_change) = 0; + + // As above, but also returns the linear filter output. + virtual void ProcessCapture(AudioBuffer* capture, + AudioBuffer* linear_output, + bool level_change) = 0; + + struct Metrics { + double echo_return_loss; + double echo_return_loss_enhancement; + int delay_ms; + }; + + // Collect current metrics from the echo controller. + virtual Metrics GetMetrics() const = 0; + + // Provides an optional external estimate of the audio buffer delay. + virtual void SetAudioBufferDelay(int delay_ms) = 0; + + // Specifies whether the capture output will be used. The purpose of this is + // to allow the echo controller to deactivate some of the processing when the + // resulting output is anyway not used, for instance when the endpoint is + // muted. + // TODO(b/177830919): Make pure virtual. + virtual void SetCaptureOutputUsage(bool capture_output_used) {} + + // Returns wheter the signal is altered. + virtual bool ActiveProcessing() const = 0; + + virtual ~EchoControl() {} +}; + +// Interface for a factory that creates EchoControllers. +class EchoControlFactory { + public: + virtual std::unique_ptr Create(int sample_rate_hz, + int num_render_channels, + int num_capture_channels) = 0; + + virtual ~EchoControlFactory() = default; +}; +} // namespace webrtc + +#endif // API_AUDIO_ECHO_CONTROL_H_ diff --git a/third_party/libwebrtc/api/audio/echo_control_gn/moz.build b/third_party/libwebrtc/api/audio/echo_control_gn/moz.build new file mode 100644 index 0000000000..6a5ce44f46 --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_control_gn/moz.build @@ -0,0 +1,209 @@ +# 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" +] + +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 + +if CONFIG["TARGET_CPU"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["TARGET_CPU"] == "arm": + + 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": + + 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": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64": + + DEFINES["_GNU_SOURCE"] = True + +Library("echo_control_gn") diff --git a/third_party/libwebrtc/api/audio/echo_detector_creator.cc b/third_party/libwebrtc/api/audio/echo_detector_creator.cc new file mode 100644 index 0000000000..15b7c51dca --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_detector_creator.cc @@ -0,0 +1,21 @@ +/* + * 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 "api/audio/echo_detector_creator.h" + +#include "api/make_ref_counted.h" +#include "modules/audio_processing/residual_echo_detector.h" + +namespace webrtc { + +rtc::scoped_refptr CreateEchoDetector() { + return rtc::make_ref_counted(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/echo_detector_creator.h b/third_party/libwebrtc/api/audio/echo_detector_creator.h new file mode 100644 index 0000000000..5ba171de97 --- /dev/null +++ b/third_party/libwebrtc/api/audio/echo_detector_creator.h @@ -0,0 +1,26 @@ +/* + * 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 API_AUDIO_ECHO_DETECTOR_CREATOR_H_ +#define API_AUDIO_ECHO_DETECTOR_CREATOR_H_ + +#include "api/scoped_refptr.h" +#include "modules/audio_processing/include/audio_processing.h" + +namespace webrtc { + +// Returns an instance of the WebRTC implementation of a residual echo detector. +// It can be provided to the webrtc::AudioProcessingBuilder to obtain the +// usual residual echo metrics. +rtc::scoped_refptr CreateEchoDetector(); + +} // namespace webrtc + +#endif // API_AUDIO_ECHO_DETECTOR_CREATOR_H_ diff --git a/third_party/libwebrtc/api/audio/test/BUILD.gn b/third_party/libwebrtc/api/audio/test/BUILD.gn new file mode 100644 index 0000000000..8d5822fa54 --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/BUILD.gn @@ -0,0 +1,29 @@ +# Copyright (c) 2018 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") +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} + +if (rtc_include_tests) { + rtc_library("audio_api_unittests") { + testonly = true + sources = [ + "audio_frame_unittest.cc", + "echo_canceller3_config_unittest.cc", + ] + deps = [ + "..:aec3_config", + "..:audio_frame_api", + "../../../modules/audio_processing:aec3_config_json", + "../../../test:test_support", + ] + } +} diff --git a/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc b/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc new file mode 100644 index 0000000000..dbf45ceabc --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/audio_frame_unittest.cc @@ -0,0 +1,136 @@ +/* + * Copyright 2018 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 "api/audio/audio_frame.h" + +#include +#include // memcmp + +#include "test/gtest.h" + +namespace webrtc { + +namespace { + +bool AllSamplesAre(int16_t sample, const AudioFrame& frame) { + const int16_t* frame_data = frame.data(); + for (size_t i = 0; i < frame.max_16bit_samples(); i++) { + if (frame_data[i] != sample) { + return false; + } + } + return true; +} + +constexpr uint32_t kTimestamp = 27; +constexpr int kSampleRateHz = 16000; +constexpr size_t kNumChannelsMono = 1; +constexpr size_t kNumChannelsStereo = 2; +constexpr size_t kNumChannels5_1 = 6; +constexpr size_t kSamplesPerChannel = kSampleRateHz / 100; + +} // namespace + +TEST(AudioFrameTest, FrameStartsMuted) { + AudioFrame frame; + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) { + AudioFrame frame; + frame.mutable_data(); + EXPECT_FALSE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, MutedFrameBufferIsZeroed) { + AudioFrame frame; + int16_t* frame_data = frame.mutable_data(); + for (size_t i = 0; i < frame.max_16bit_samples(); i++) { + frame_data[i] = 17; + } + ASSERT_TRUE(AllSamplesAre(17, frame)); + frame.Mute(); + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UpdateFrameMono) { + AudioFrame frame; + int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17}; + frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz, + AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannelsMono); + + EXPECT_EQ(kTimestamp, frame.timestamp_); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz()); + EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_); + EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_); + EXPECT_EQ(kNumChannelsMono, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame.channel_layout()); + + EXPECT_FALSE(frame.muted()); + EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples))); + + frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + EXPECT_TRUE(frame.muted()); + EXPECT_TRUE(AllSamplesAre(0, frame)); +} + +TEST(AudioFrameTest, UpdateFrameMultiChannel) { + AudioFrame frame; + frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsStereo); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kNumChannelsStereo, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame.channel_layout()); + + frame.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannels5_1); + EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel()); + EXPECT_EQ(kNumChannels5_1, frame.num_channels()); + EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame.channel_layout()); +} + +TEST(AudioFrameTest, CopyFrom) { + AudioFrame frame1; + AudioFrame frame2; + + int16_t samples[kNumChannelsMono * kSamplesPerChannel] = {17}; + frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz, + AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + frame1.CopyFrom(frame2); + + EXPECT_EQ(frame2.timestamp_, frame1.timestamp_); + EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_); + EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_); + EXPECT_EQ(frame2.speech_type_, frame1.speech_type_); + EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_); + EXPECT_EQ(frame2.num_channels_, frame1.num_channels_); + + EXPECT_EQ(frame2.muted(), frame1.muted()); + EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples))); + + frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel, + kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive, + kNumChannelsMono); + frame1.CopyFrom(frame2); + + EXPECT_EQ(frame2.muted(), frame1.muted()); + EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples))); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc new file mode 100644 index 0000000000..da0255806e --- /dev/null +++ b/third_party/libwebrtc/api/audio/test/echo_canceller3_config_unittest.cc @@ -0,0 +1,46 @@ +/* + * Copyright 2018 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 "api/audio/echo_canceller3_config.h" + +#include "modules/audio_processing/test/echo_canceller3_config_json.h" +#include "test/gtest.h" + +namespace webrtc { + +TEST(EchoCanceller3Config, ValidConfigIsNotModified) { + EchoCanceller3Config config; + EXPECT_TRUE(EchoCanceller3Config::Validate(&config)); + EchoCanceller3Config default_config; + EXPECT_EQ(Aec3ConfigToJsonString(config), + Aec3ConfigToJsonString(default_config)); +} + +TEST(EchoCanceller3Config, InvalidConfigIsCorrected) { + // Change a parameter and validate. + EchoCanceller3Config config; + config.echo_model.min_noise_floor_power = -1600000.f; + EXPECT_FALSE(EchoCanceller3Config::Validate(&config)); + EXPECT_GE(config.echo_model.min_noise_floor_power, 0.f); + // Verify remaining parameters are unchanged. + EchoCanceller3Config default_config; + config.echo_model.min_noise_floor_power = + default_config.echo_model.min_noise_floor_power; + EXPECT_EQ(Aec3ConfigToJsonString(config), + Aec3ConfigToJsonString(default_config)); +} + +TEST(EchoCanceller3Config, ValidatedConfigsAreValid) { + EchoCanceller3Config config; + config.delay.down_sampling_factor = 983; + EXPECT_FALSE(EchoCanceller3Config::Validate(&config)); + EXPECT_TRUE(EchoCanceller3Config::Validate(&config)); +} +} // namespace webrtc -- cgit v1.2.3