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/transport/rtp/BUILD.gn | 38 ++++ .../api/transport/rtp/dependency_descriptor.cc | 54 +++++ .../api/transport/rtp/dependency_descriptor.h | 129 ++++++++++++ .../rtp/dependency_descriptor_gn/moz.build | 225 +++++++++++++++++++++ .../libwebrtc/api/transport/rtp/rtp_source.h | 108 ++++++++++ .../api/transport/rtp/rtp_source_gn/moz.build | 216 ++++++++++++++++++++ 6 files changed, 770 insertions(+) create mode 100644 third_party/libwebrtc/api/transport/rtp/BUILD.gn create mode 100644 third_party/libwebrtc/api/transport/rtp/dependency_descriptor.cc create mode 100644 third_party/libwebrtc/api/transport/rtp/dependency_descriptor.h create mode 100644 third_party/libwebrtc/api/transport/rtp/dependency_descriptor_gn/moz.build create mode 100644 third_party/libwebrtc/api/transport/rtp/rtp_source.h create mode 100644 third_party/libwebrtc/api/transport/rtp/rtp_source_gn/moz.build (limited to 'third_party/libwebrtc/api/transport/rtp') diff --git a/third_party/libwebrtc/api/transport/rtp/BUILD.gn b/third_party/libwebrtc/api/transport/rtp/BUILD.gn new file mode 100644 index 0000000000..6f2a15e0e9 --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/BUILD.gn @@ -0,0 +1,38 @@ +# 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. + +import("../../../webrtc.gni") + +rtc_source_set("rtp_source") { + visibility = [ "*" ] + sources = [ "rtp_source.h" ] + deps = [ + "../../../api:rtp_headers", + "../../../api/units:time_delta", + "../../../api/units:timestamp", + "../../../rtc_base:checks", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] +} + +rtc_source_set("dependency_descriptor") { + visibility = [ "*" ] + sources = [ + "dependency_descriptor.cc", + "dependency_descriptor.h", + ] + deps = [ + "../../../rtc_base:checks", + "../../video:render_resolution", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/container:inlined_vector", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} diff --git a/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.cc b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.cc new file mode 100644 index 0000000000..e784853cd6 --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.cc @@ -0,0 +1,54 @@ +/* + * 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/transport/rtp/dependency_descriptor.h" + +#include "absl/container/inlined_vector.h" +#include "absl/strings/string_view.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +constexpr int DependencyDescriptor::kMaxSpatialIds; +constexpr int DependencyDescriptor::kMaxTemporalIds; +constexpr int DependencyDescriptor::kMaxTemplates; +constexpr int DependencyDescriptor::kMaxDecodeTargets; + +namespace webrtc_impl { + +absl::InlinedVector StringToDecodeTargetIndications( + absl::string_view symbols) { + absl::InlinedVector dtis; + dtis.reserve(symbols.size()); + for (char symbol : symbols) { + DecodeTargetIndication indication; + switch (symbol) { + case '-': + indication = DecodeTargetIndication::kNotPresent; + break; + case 'D': + indication = DecodeTargetIndication::kDiscardable; + break; + case 'R': + indication = DecodeTargetIndication::kRequired; + break; + case 'S': + indication = DecodeTargetIndication::kSwitch; + break; + default: + RTC_DCHECK_NOTREACHED(); + } + dtis.push_back(indication); + } + return dtis; +} + +} // namespace webrtc_impl +} // namespace webrtc diff --git a/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.h b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.h new file mode 100644 index 0000000000..0db600918e --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor.h @@ -0,0 +1,129 @@ +/* + * 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_TRANSPORT_RTP_DEPENDENCY_DESCRIPTOR_H_ +#define API_TRANSPORT_RTP_DEPENDENCY_DESCRIPTOR_H_ + +#include + +#include +#include +#include + +#include "absl/container/inlined_vector.h" +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "api/video/render_resolution.h" + +namespace webrtc { +// Structures to build and parse dependency descriptor as described in +// https://aomediacodec.github.io/av1-rtp-spec/#dependency-descriptor-rtp-header-extension + +// Relationship of a frame to a Decode target. +enum class DecodeTargetIndication { + kNotPresent = 0, // DecodeTargetInfo symbol '-' + kDiscardable = 1, // DecodeTargetInfo symbol 'D' + kSwitch = 2, // DecodeTargetInfo symbol 'S' + kRequired = 3 // DecodeTargetInfo symbol 'R' +}; + +struct FrameDependencyTemplate { + // Setters are named briefly to chain them when building the template. + FrameDependencyTemplate& S(int spatial_layer); + FrameDependencyTemplate& T(int temporal_layer); + FrameDependencyTemplate& Dtis(absl::string_view dtis); + FrameDependencyTemplate& FrameDiffs(std::initializer_list diffs); + FrameDependencyTemplate& ChainDiffs(std::initializer_list diffs); + + friend bool operator==(const FrameDependencyTemplate& lhs, + const FrameDependencyTemplate& rhs) { + return lhs.spatial_id == rhs.spatial_id && + lhs.temporal_id == rhs.temporal_id && + lhs.decode_target_indications == rhs.decode_target_indications && + lhs.frame_diffs == rhs.frame_diffs && + lhs.chain_diffs == rhs.chain_diffs; + } + + int spatial_id = 0; + int temporal_id = 0; + absl::InlinedVector decode_target_indications; + absl::InlinedVector frame_diffs; + absl::InlinedVector chain_diffs; +}; + +struct FrameDependencyStructure { + friend bool operator==(const FrameDependencyStructure& lhs, + const FrameDependencyStructure& rhs) { + return lhs.num_decode_targets == rhs.num_decode_targets && + lhs.num_chains == rhs.num_chains && + lhs.decode_target_protected_by_chain == + rhs.decode_target_protected_by_chain && + lhs.resolutions == rhs.resolutions && lhs.templates == rhs.templates; + } + + int structure_id = 0; + int num_decode_targets = 0; + int num_chains = 0; + // If chains are used (num_chains > 0), maps decode target index into index of + // the chain protecting that target. + absl::InlinedVector decode_target_protected_by_chain; + absl::InlinedVector resolutions; + std::vector templates; +}; + +struct DependencyDescriptor { + static constexpr int kMaxSpatialIds = 4; + static constexpr int kMaxTemporalIds = 8; + static constexpr int kMaxDecodeTargets = 32; + static constexpr int kMaxTemplates = 64; + + bool first_packet_in_frame = true; + bool last_packet_in_frame = true; + int frame_number = 0; + FrameDependencyTemplate frame_dependencies; + absl::optional resolution; + absl::optional active_decode_targets_bitmask; + std::unique_ptr attached_structure; +}; + +// Below are implementation details. +namespace webrtc_impl { +absl::InlinedVector StringToDecodeTargetIndications( + absl::string_view indication_symbols); +} // namespace webrtc_impl + +inline FrameDependencyTemplate& FrameDependencyTemplate::S(int spatial_layer) { + this->spatial_id = spatial_layer; + return *this; +} +inline FrameDependencyTemplate& FrameDependencyTemplate::T(int temporal_layer) { + this->temporal_id = temporal_layer; + return *this; +} +inline FrameDependencyTemplate& FrameDependencyTemplate::Dtis( + absl::string_view dtis) { + this->decode_target_indications = + webrtc_impl::StringToDecodeTargetIndications(dtis); + return *this; +} +inline FrameDependencyTemplate& FrameDependencyTemplate::FrameDiffs( + std::initializer_list diffs) { + this->frame_diffs.assign(diffs.begin(), diffs.end()); + return *this; +} +inline FrameDependencyTemplate& FrameDependencyTemplate::ChainDiffs( + std::initializer_list diffs) { + this->chain_diffs.assign(diffs.begin(), diffs.end()); + return *this; +} + +} // namespace webrtc + +#endif // API_TRANSPORT_RTP_DEPENDENCY_DESCRIPTOR_H_ diff --git a/third_party/libwebrtc/api/transport/rtp/dependency_descriptor_gn/moz.build b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor_gn/moz.build new file mode 100644 index 0000000000..a2fe75e60e --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/dependency_descriptor_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/transport/rtp/dependency_descriptor.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("dependency_descriptor_gn") diff --git a/third_party/libwebrtc/api/transport/rtp/rtp_source.h b/third_party/libwebrtc/api/transport/rtp/rtp_source.h new file mode 100644 index 0000000000..11149f5421 --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/rtp_source.h @@ -0,0 +1,108 @@ +/* + * Copyright 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_TRANSPORT_RTP_RTP_SOURCE_H_ +#define API_TRANSPORT_RTP_RTP_SOURCE_H_ + +#include + +#include "absl/types/optional.h" +#include "api/rtp_headers.h" +#include "api/units/time_delta.h" +#include "api/units/timestamp.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +enum class RtpSourceType { + SSRC, + CSRC, +}; + +class RtpSource { + public: + struct Extensions { + absl::optional audio_level; + + // Fields from the Absolute Capture Time header extension: + // http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time + absl::optional absolute_capture_time; + + // Clock offset between the local clock and the capturer's clock. + // Do not confuse with `AbsoluteCaptureTime::estimated_capture_clock_offset` + // which instead represents the clock offset between a remote sender and the + // capturer. The following holds: + // Capture's NTP Clock = Local NTP Clock + Local-Capture Clock Offset + absl::optional local_capture_clock_offset; + }; + + RtpSource() = delete; + + RtpSource(Timestamp timestamp, + uint32_t source_id, + RtpSourceType source_type, + uint32_t rtp_timestamp, + const RtpSource::Extensions& extensions) + : timestamp_(timestamp), + source_id_(source_id), + source_type_(source_type), + extensions_(extensions), + rtp_timestamp_(rtp_timestamp) {} + + RtpSource(const RtpSource&) = default; + RtpSource& operator=(const RtpSource&) = default; + ~RtpSource() = default; + + Timestamp timestamp() const { return timestamp_; } + + // The identifier of the source can be the CSRC or the SSRC. + uint32_t source_id() const { return source_id_; } + + // The source can be either a contributing source or a synchronization source. + RtpSourceType source_type() const { return source_type_; } + + absl::optional audio_level() const { + return extensions_.audio_level; + } + + void set_audio_level(const absl::optional& level) { + extensions_.audio_level = level; + } + + uint32_t rtp_timestamp() const { return rtp_timestamp_; } + + absl::optional absolute_capture_time() const { + return extensions_.absolute_capture_time; + } + + absl::optional local_capture_clock_offset() const { + return extensions_.local_capture_clock_offset; + } + + bool operator==(const RtpSource& o) const { + return timestamp_ == o.timestamp() && source_id_ == o.source_id() && + source_type_ == o.source_type() && + extensions_.audio_level == o.extensions_.audio_level && + extensions_.absolute_capture_time == + o.extensions_.absolute_capture_time && + rtp_timestamp_ == o.rtp_timestamp(); + } + + private: + Timestamp timestamp_; + uint32_t source_id_; + RtpSourceType source_type_; + RtpSource::Extensions extensions_; + uint32_t rtp_timestamp_; +}; + +} // namespace webrtc + +#endif // API_TRANSPORT_RTP_RTP_SOURCE_H_ diff --git a/third_party/libwebrtc/api/transport/rtp/rtp_source_gn/moz.build b/third_party/libwebrtc/api/transport/rtp/rtp_source_gn/moz.build new file mode 100644 index 0000000000..9a7b0b69bf --- /dev/null +++ b/third_party/libwebrtc/api/transport/rtp/rtp_source_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("rtp_source_gn") -- cgit v1.2.3