diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/g722 | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/g722')
9 files changed, 967 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn new file mode 100644 index 0000000000..af13ac3de3 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn @@ -0,0 +1,62 @@ +# Copyright (c) 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. + +import("../../../webrtc.gni") +if (is_android) { + import("//build/config/android/config.gni") + import("//build/config/android/rules.gni") +} + +rtc_source_set("audio_encoder_g722_config") { + visibility = [ "*" ] + sources = [ "audio_encoder_g722_config.h" ] + deps = [ "..:audio_codecs_api" ] +} + +rtc_library("audio_encoder_g722") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_encoder_g722.cc", + "audio_encoder_g722.h", + ] + deps = [ + ":audio_encoder_g722_config", + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:g722", + "../../../rtc_base:safe_conversions", + "../../../rtc_base:safe_minmax", + "../../../rtc_base:stringutils", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("audio_decoder_g722") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_g722.cc", + "audio_decoder_g722.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:g722", + "../../../rtc_base:safe_conversions", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc new file mode 100644 index 0000000000..ed7163471a --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc @@ -0,0 +1,56 @@ +/* + * Copyright (c) 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. + */ + +#include "api/audio_codecs/g722/audio_decoder_g722.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h" +#include "rtc_base/numerics/safe_conversions.h" + +namespace webrtc { + +absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig( + const SdpAudioFormat& format) { + if (absl::EqualsIgnoreCase(format.name, "G722") && + format.clockrate_hz == 8000 && + (format.num_channels == 1 || format.num_channels == 2)) { + return Config{rtc::dchecked_cast<int>(format.num_channels)}; + } + return absl::nullopt; +} + +void AudioDecoderG722::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}}); +} + +std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder( + Config config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + switch (config.num_channels) { + case 1: + return std::make_unique<AudioDecoderG722Impl>(); + case 2: + return std::make_unique<AudioDecoderG722StereoImpl>(); + default: + RTC_DCHECK_NOTREACHED(); + return nullptr; + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h new file mode 100644 index 0000000000..6f7b253039 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 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_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_ +#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_ + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_decoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// G722 decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct RTC_EXPORT AudioDecoderG722 { + struct Config { + bool IsOk() const { return num_channels == 1 || num_channels == 2; } + int num_channels; + }; + static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs); + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + Config config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_gn/moz.build new file mode 100644 index 0000000000..77003c77a9 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_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_codecs/g722/audio_decoder_g722.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_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 + +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 += [ + "winmm" + ] + +if CONFIG["CPU_ARCH"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "arm": + + CXXFLAGS += [ + "-mfpu=neon" + ] + + 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": + + CXXFLAGS += [ + "-msse2" + ] + + 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": + + CXXFLAGS += [ + "-msse2" + ] + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +Library("audio_decoder_g722_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc new file mode 100644 index 0000000000..56a6c4da6a --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc @@ -0,0 +1,74 @@ +/* + * Copyright (c) 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. + */ + +#include "api/audio_codecs/g722/audio_encoder_g722.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { + +absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig( + const SdpAudioFormat& format) { + if (!absl::EqualsIgnoreCase(format.name, "g722") || + format.clockrate_hz != 8000) { + return absl::nullopt; + } + + AudioEncoderG722Config config; + config.num_channels = rtc::checked_cast<int>(format.num_channels); + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + auto ptime = rtc::StringToNumber<int>(ptime_iter->second); + if (ptime && *ptime > 0) { + const int whole_packets = *ptime / 10; + config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60); + } + } + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; +} + +void AudioEncoderG722::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + const SdpAudioFormat fmt = {"G722", 8000, 1}; + const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); + specs->push_back({fmt, info}); +} + +AudioCodecInfo AudioEncoderG722::QueryAudioEncoder( + const AudioEncoderG722Config& config) { + RTC_DCHECK(config.IsOk()); + return {16000, rtc::dchecked_cast<size_t>(config.num_channels), + 64000 * config.num_channels}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder( + const AudioEncoderG722Config& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioEncoderG722Impl>(config, payload_type); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h new file mode 100644 index 0000000000..78ceddd1e9 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h @@ -0,0 +1,44 @@ +/* + * Copyright (c) 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_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_ +#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_ + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/audio_codec_pair_id.h" +#include "api/audio_codecs/audio_encoder.h" +#include "api/audio_codecs/audio_format.h" +#include "api/audio_codecs/g722/audio_encoder_g722_config.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// G722 encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderG722 { + using Config = AudioEncoderG722Config; + static absl::optional<AudioEncoderG722Config> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const AudioEncoderG722Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h new file mode 100644 index 0000000000..f3f3a9f016 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h @@ -0,0 +1,29 @@ +/* + * Copyright (c) 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_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ +#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ + +#include "api/audio_codecs/audio_encoder.h" + +namespace webrtc { + +struct AudioEncoderG722Config { + bool IsOk() const { + return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 && + num_channels <= AudioEncoder::kMaxNumberOfChannels; + } + int frame_size_ms = 20; + int num_channels = 1; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_gn/moz.build new file mode 100644 index 0000000000..41e1e248c5 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_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_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 + +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 += [ + "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("audio_encoder_g722_config_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_gn/moz.build new file mode 100644 index 0000000000..c3beba6cdb --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_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_codecs/g722/audio_encoder_g722.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_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 + +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 += [ + "winmm" + ] + +if CONFIG["CPU_ARCH"] == "aarch64": + + DEFINES["WEBRTC_ARCH_ARM64"] = True + DEFINES["WEBRTC_HAS_NEON"] = True + +if CONFIG["CPU_ARCH"] == "arm": + + CXXFLAGS += [ + "-mfpu=neon" + ] + + 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": + + CXXFLAGS += [ + "-msse2" + ] + + 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": + + CXXFLAGS += [ + "-msse2" + ] + + DEFINES["_GNU_SOURCE"] = True + +if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux": + + DEFINES["_GNU_SOURCE"] = True + +Library("audio_encoder_g722_gn") |