summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/api/call
diff options
context:
space:
mode:
Diffstat (limited to '')
-rw-r--r--third_party/libwebrtc/api/call/audio_sink.h48
-rw-r--r--third_party/libwebrtc/api/call/bitrate_allocation.h45
-rw-r--r--third_party/libwebrtc/api/call/call_factory_interface.h38
-rw-r--r--third_party/libwebrtc/api/call/transport.cc23
-rw-r--r--third_party/libwebrtc/api/call/transport.h54
-rw-r--r--third_party/libwebrtc/api/call_api_gn/moz.build201
-rw-r--r--third_party/libwebrtc/api/callfactory_api_gn/moz.build216
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")
generated by cgit v1.2.3 (git 2.39.1) at 2024-09-04 00:47:27 +0000