diff options
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/call/audio_sink.h | 48 | ||||
-rw-r--r-- | third_party/libwebrtc/api/call/bitrate_allocation.h | 45 | ||||
-rw-r--r-- | third_party/libwebrtc/api/call/call_factory_interface.h | 38 | ||||
-rw-r--r-- | third_party/libwebrtc/api/call/transport.cc | 23 | ||||
-rw-r--r-- | third_party/libwebrtc/api/call/transport.h | 54 | ||||
-rw-r--r-- | third_party/libwebrtc/api/call_api_gn/moz.build | 201 | ||||
-rw-r--r-- | third_party/libwebrtc/api/callfactory_api_gn/moz.build | 216 |
7 files changed, 625 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/call/audio_sink.h b/third_party/libwebrtc/api/call/audio_sink.h new file mode 100644 index 0000000000..fec26593a6 --- /dev/null +++ b/third_party/libwebrtc/api/call/audio_sink.h @@ -0,0 +1,48 @@ +/* + * Copyright (c) 2015 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_CALL_AUDIO_SINK_H_ +#define API_CALL_AUDIO_SINK_H_ + +#include <stddef.h> +#include <stdint.h> + +namespace webrtc { + +// Represents a simple push audio sink. +class AudioSinkInterface { + public: + virtual ~AudioSinkInterface() {} + + struct Data { + Data(const int16_t* data, + size_t samples_per_channel, + int sample_rate, + size_t channels, + uint32_t timestamp) + : data(data), + samples_per_channel(samples_per_channel), + sample_rate(sample_rate), + channels(channels), + timestamp(timestamp) {} + + const int16_t* data; // The actual 16bit audio data. + size_t samples_per_channel; // Number of frames in the buffer. + int sample_rate; // Sample rate in Hz. + size_t channels; // Number of channels in the audio data. + uint32_t timestamp; // The RTP timestamp of the first sample. + }; + + virtual void OnData(const Data& audio) = 0; +}; + +} // namespace webrtc + +#endif // API_CALL_AUDIO_SINK_H_ diff --git a/third_party/libwebrtc/api/call/bitrate_allocation.h b/third_party/libwebrtc/api/call/bitrate_allocation.h new file mode 100644 index 0000000000..4b4e5e7ae1 --- /dev/null +++ b/third_party/libwebrtc/api/call/bitrate_allocation.h @@ -0,0 +1,45 @@ +/* + * 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. + */ +#ifndef API_CALL_BITRATE_ALLOCATION_H_ +#define API_CALL_BITRATE_ALLOCATION_H_ + +#include "api/units/data_rate.h" +#include "api/units/time_delta.h" + +namespace webrtc { + +// BitrateAllocationUpdate provides information to allocated streams about their +// bitrate allocation. It originates from the BitrateAllocater class and is +// propagated from there. +struct BitrateAllocationUpdate { + // The allocated target bitrate. Media streams should produce this amount of + // data. (Note that this may include packet overhead depending on + // configuration.) + DataRate target_bitrate = DataRate::Zero(); + // The allocated part of the estimated link capacity. This is more stable than + // the target as it is based on the underlying link capacity estimate. This + // should be used to change encoder configuration when the cost of change is + // high. + DataRate stable_target_bitrate = DataRate::Zero(); + // Predicted packet loss ratio. + double packet_loss_ratio = 0; + // Predicted round trip time. + TimeDelta round_trip_time = TimeDelta::PlusInfinity(); + // `bwe_period` is deprecated, use `stable_target_bitrate` allocation instead. + TimeDelta bwe_period = TimeDelta::PlusInfinity(); + // Congestion window pushback bitrate reduction fraction. Used in + // VideoStreamEncoder to reduce the bitrate by the given fraction + // by dropping frames. + double cwnd_reduce_ratio = 0; +}; + +} // namespace webrtc + +#endif // API_CALL_BITRATE_ALLOCATION_H_ diff --git a/third_party/libwebrtc/api/call/call_factory_interface.h b/third_party/libwebrtc/api/call/call_factory_interface.h new file mode 100644 index 0000000000..6051409cc3 --- /dev/null +++ b/third_party/libwebrtc/api/call/call_factory_interface.h @@ -0,0 +1,38 @@ +/* + * Copyright 2017 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_CALL_CALL_FACTORY_INTERFACE_H_ +#define API_CALL_CALL_FACTORY_INTERFACE_H_ + +#include <memory> + +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// These classes are not part of the API, and are treated as opaque pointers. +class Call; +struct CallConfig; + +// This interface exists to allow webrtc to be optionally built without media +// support (i.e., if only being used for data channels). PeerConnectionFactory +// is constructed with a CallFactoryInterface, which may or may not be null. +class CallFactoryInterface { + public: + virtual ~CallFactoryInterface() {} + + virtual Call* CreateCall(const CallConfig& config) = 0; +}; + +RTC_EXPORT std::unique_ptr<CallFactoryInterface> CreateCallFactory(); + +} // namespace webrtc + +#endif // API_CALL_CALL_FACTORY_INTERFACE_H_ diff --git a/third_party/libwebrtc/api/call/transport.cc b/third_party/libwebrtc/api/call/transport.cc new file mode 100644 index 0000000000..bcadc762de --- /dev/null +++ b/third_party/libwebrtc/api/call/transport.cc @@ -0,0 +1,23 @@ +/* + * 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/call/transport.h" + +#include <cstdint> + +namespace webrtc { + +PacketOptions::PacketOptions() = default; + +PacketOptions::PacketOptions(const PacketOptions&) = default; + +PacketOptions::~PacketOptions() = default; + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/call/transport.h b/third_party/libwebrtc/api/call/transport.h new file mode 100644 index 0000000000..8bff28825d --- /dev/null +++ b/third_party/libwebrtc/api/call/transport.h @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2013 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_CALL_TRANSPORT_H_ +#define API_CALL_TRANSPORT_H_ + +#include <stddef.h> +#include <stdint.h> + +#include "api/ref_counted_base.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// TODO(holmer): Look into unifying this with the PacketOptions in +// asyncpacketsocket.h. +struct PacketOptions { + PacketOptions(); + PacketOptions(const PacketOptions&); + ~PacketOptions(); + + // A 16 bits positive id. Negative ids are invalid and should be interpreted + // as packet_id not being set. + int packet_id = -1; + // Additional data bound to the RTP packet for use in application code, + // outside of WebRTC. + rtc::scoped_refptr<rtc::RefCountedBase> additional_data; + // Whether this is a retransmission of an earlier packet. + bool is_retransmit = false; + bool included_in_feedback = false; + bool included_in_allocation = false; +}; + +class Transport { + public: + virtual bool SendRtp(const uint8_t* packet, + size_t length, + const PacketOptions& options) = 0; + virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0; + + protected: + virtual ~Transport() {} +}; + +} // namespace webrtc + +#endif // API_CALL_TRANSPORT_H_ diff --git a/third_party/libwebrtc/api/call_api_gn/moz.build b/third_party/libwebrtc/api/call_api_gn/moz.build new file mode 100644 index 0000000000..e446d82280 --- /dev/null +++ b/third_party/libwebrtc/api/call_api_gn/moz.build @@ -0,0 +1,201 @@ +# 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_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_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_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["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["CPU_ARCH"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "arm": + + DEFINES["WEBRTC_ARCH_ARM"] = True + DEFINES["WEBRTC_ARCH_ARM_V7"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "mips32": + + DEFINES["MIPS32_LE"] = True + DEFINES["MIPS_FPU_LE"] = True + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "mips64": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android": + + OS_LIBS += [ + "android_support", + "unwind" + ] + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android": + + OS_LIBS += [ + "android_support" + ] + +if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +Library("call_api_gn") diff --git a/third_party/libwebrtc/api/callfactory_api_gn/moz.build b/third_party/libwebrtc/api/callfactory_api_gn/moz.build new file mode 100644 index 0000000000..b0f2bbbf35 --- /dev/null +++ b/third_party/libwebrtc/api/callfactory_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_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_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_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["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["CPU_ARCH"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "arm": + + DEFINES["WEBRTC_ARCH_ARM"] = True + DEFINES["WEBRTC_ARCH_ARM_V7"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "mips32": + + DEFINES["MIPS32_LE"] = True + DEFINES["MIPS_FPU_LE"] = True + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "mips64": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86": + + DEFINES["WEBRTC_ENABLE_AVX2"] = True + +if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android": + + OS_LIBS += [ + "android_support", + "unwind" + ] + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android": + + OS_LIBS += [ + "android_support" + ] + +if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +Library("callfactory_api_gn") |