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Diffstat (limited to '')
80 files changed, 8955 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/audio_codecs/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/BUILD.gn new file mode 100644 index 0000000000..82ed31a5da --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/BUILD.gn @@ -0,0 +1,144 @@ +# 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_library("audio_codecs_api") { + visibility = [ "*" ] + sources = [ + "audio_codec_pair_id.cc", + "audio_codec_pair_id.h", + "audio_decoder.cc", + "audio_decoder.h", + "audio_decoder_factory.h", + "audio_decoder_factory_template.h", + "audio_encoder.cc", + "audio_encoder.h", + "audio_encoder_factory.h", + "audio_encoder_factory_template.h", + "audio_format.cc", + "audio_format.h", + ] + deps = [ + "..:array_view", + "..:bitrate_allocation", + "..:make_ref_counted", + "..:scoped_refptr", + "../../api:field_trials_view", + "../../rtc_base:buffer", + "../../rtc_base:checks", + "../../rtc_base:event_tracer", + "../../rtc_base:refcount", + "../../rtc_base:sanitizer", + "../../rtc_base/system:rtc_export", + "../units:time_delta", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/base:core_headers", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("builtin_audio_decoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "builtin_audio_decoder_factory.cc", + "builtin_audio_decoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "L16:audio_decoder_L16", + "g711:audio_decoder_g711", + "g722:audio_decoder_g722", + ] + defines = [] + if (rtc_include_ilbc) { + deps += [ "ilbc:audio_decoder_ilbc" ] + defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ] + } + if (rtc_include_opus) { + deps += [ + "opus:audio_decoder_multiopus", + "opus:audio_decoder_opus", + ] + defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] + } +} + +rtc_library("builtin_audio_encoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "builtin_audio_encoder_factory.cc", + "builtin_audio_encoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "L16:audio_encoder_L16", + "g711:audio_encoder_g711", + "g722:audio_encoder_g722", + ] + defines = [] + if (rtc_include_ilbc) { + deps += [ "ilbc:audio_encoder_ilbc" ] + defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ] + } + if (rtc_include_opus) { + deps += [ + "opus:audio_encoder_multiopus", + "opus:audio_encoder_opus", + ] + defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] + } else { + defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] + } +} + +rtc_library("opus_audio_decoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "opus_audio_decoder_factory.cc", + "opus_audio_decoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "opus:audio_decoder_multiopus", + "opus:audio_decoder_opus", + ] +} + +rtc_library("opus_audio_encoder_factory") { + visibility = [ "*" ] + allow_poison = [ "audio_codecs" ] + sources = [ + "opus_audio_encoder_factory.cc", + "opus_audio_encoder_factory.h", + ] + deps = [ + ":audio_codecs_api", + "..:scoped_refptr", + "opus:audio_encoder_multiopus", + "opus:audio_encoder_opus", + ] +} diff --git a/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn new file mode 100644 index 0000000000..41e9eb42d8 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn @@ -0,0 +1,55 @@ +# 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_library("audio_encoder_L16") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_encoder_L16.cc", + "audio_encoder_L16.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:pcm16b", + "../../../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_L16") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_L16.cc", + "audio_decoder_L16.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:pcm16b", + "../../../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/L16/audio_decoder_L16.cc b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc new file mode 100644 index 0000000000..a03abe26f7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc @@ -0,0 +1,49 @@ +/* + * 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/L16/audio_decoder_L16.h" + +#include <memory> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" +#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h" +#include "rtc_base/numerics/safe_conversions.h" + +namespace webrtc { + +absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig( + const SdpAudioFormat& format) { + Config config; + config.sample_rate_hz = format.clockrate_hz; + config.num_channels = rtc::checked_cast<int>(format.num_channels); + if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) { + return config; + } + return absl::nullopt; +} + +void AudioDecoderL16::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + Pcm16BAppendSupportedCodecSpecs(specs); +} + +std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder( + const Config& config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + return nullptr; + } + return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz, + config.num_channels); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h new file mode 100644 index 0000000000..5a01b7dc01 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h @@ -0,0 +1,49 @@ +/* + * 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_L16_AUDIO_DECODER_L16_H_ +#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_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 { + +// L16 decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct RTC_EXPORT AudioDecoderL16 { + struct Config { + bool IsOk() const { + return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || + sample_rate_hz == 32000 || sample_rate_hz == 48000) && + (num_channels >= 1 && + num_channels <= AudioDecoder::kMaxNumberOfChannels); + } + int sample_rate_hz = 8000; + int num_channels = 1; + }; + static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs); + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const Config& config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_gn/moz.build new file mode 100644 index 0000000000..87335c298d --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_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/L16/audio_decoder_L16.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_L16_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc new file mode 100644 index 0000000000..20259b9ad8 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc @@ -0,0 +1,76 @@ +/* + * 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/L16/audio_encoder_L16.h" + +#include <memory> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" +#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.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<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig( + const SdpAudioFormat& format) { + if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + Config config; + config.sample_rate_hz = format.clockrate_hz; + config.num_channels = rtc::dchecked_cast<int>(format.num_channels); + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + const auto ptime = rtc::StringToNumber<int>(ptime_iter->second); + if (ptime && *ptime > 0) { + config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60); + } + } + if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) { + return config; + } + return absl::nullopt; +} + +void AudioEncoderL16::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + Pcm16BAppendSupportedCodecSpecs(specs); +} + +AudioCodecInfo AudioEncoderL16::QueryAudioEncoder( + const AudioEncoderL16::Config& config) { + RTC_DCHECK(config.IsOk()); + return {config.sample_rate_hz, + rtc::dchecked_cast<size_t>(config.num_channels), + config.sample_rate_hz * config.num_channels * 16}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder( + const AudioEncoderL16::Config& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + AudioEncoderPcm16B::Config c; + c.sample_rate_hz = config.sample_rate_hz; + c.num_channels = config.num_channels; + c.frame_size_ms = config.frame_size_ms; + c.payload_type = payload_type; + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioEncoderPcm16B>(c); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h new file mode 100644 index 0000000000..47509849de --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h @@ -0,0 +1,54 @@ +/* + * 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_L16_AUDIO_ENCODER_L16_H_ +#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_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/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// L16 encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderL16 { + struct Config { + bool IsOk() const { + return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || + sample_rate_hz == 32000 || sample_rate_hz == 48000) && + num_channels >= 1 && + num_channels <= AudioEncoder::kMaxNumberOfChannels && + frame_size_ms > 0 && frame_size_ms <= 120 && + frame_size_ms % 10 == 0; + } + int sample_rate_hz = 8000; + int num_channels = 1; + int frame_size_ms = 10; + }; + static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_gn/moz.build new file mode 100644 index 0000000000..49e0d546f1 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_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/L16/audio_encoder_L16.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_L16_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/OWNERS b/third_party/libwebrtc/api/audio_codecs/OWNERS new file mode 100644 index 0000000000..77b414abc3 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/OWNERS @@ -0,0 +1,3 @@ +alessiob@webrtc.org +henrik.lundin@webrtc.org +jakobi@webrtc.org diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc new file mode 100644 index 0000000000..6cb51ed6b7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc @@ -0,0 +1,91 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/audio_codec_pair_id.h" + +#include <atomic> +#include <cstdint> + +#include "rtc_base/checks.h" + +namespace webrtc { + +namespace { + +// Returns a new value that it has never returned before. You may call it at +// most 2^63 times in the lifetime of the program. Note: The returned values +// may be easily predictable. +uint64_t GetNextId() { + static std::atomic<uint64_t> next_id(0); + + // Atomically increment `next_id`, and return the previous value. Relaxed + // memory order is sufficient, since all we care about is that different + // callers return different values. + const uint64_t new_id = next_id.fetch_add(1, std::memory_order_relaxed); + + // This check isn't atomic with the increment, so if we start 2^63 + 1 + // invocations of GetNextId() in parallel, the last one to do the atomic + // increment could return the ID 0 before any of the others had time to + // trigger this DCHECK. We blithely assume that this won't happen. + RTC_DCHECK_LT(new_id, uint64_t{1} << 63) << "Used up all ID values"; + + return new_id; +} + +// Make an integer ID more unpredictable. This is a 1:1 mapping, so you can +// feed it any value, but the idea is that you can feed it a sequence such as +// 0, 1, 2, ... and get a new sequence that isn't as trivially predictable, so +// that users won't rely on it being consecutive or increasing or anything like +// that. +constexpr uint64_t ObfuscateId(uint64_t id) { + // Any nonzero coefficient that's relatively prime to 2^64 (that is, any odd + // number) and any constant will give a 1:1 mapping. These high-entropy + // values will prevent the sequence from being trivially predictable. + // + // Both the multiplication and the addition going to overflow almost always, + // but that's fine---we *want* arithmetic mod 2^64. + return uint64_t{0x85fdb20e1294309a} + uint64_t{0xc516ef5c37462469} * id; +} + +// The first ten values. Verified against the Python function +// +// def f(n): +// return (0x85fdb20e1294309a + 0xc516ef5c37462469 * n) % 2**64 +// +// Callers should obviously not depend on these exact values... +// +// (On Visual C++, we have to disable warning C4307 (integral constant +// overflow), even though unsigned integers have perfectly well-defined +// overflow behavior.) +#ifdef _MSC_VER +#pragma warning(push) +#pragma warning(disable : 4307) +#endif +static_assert(ObfuscateId(0) == uint64_t{0x85fdb20e1294309a}, ""); +static_assert(ObfuscateId(1) == uint64_t{0x4b14a16a49da5503}, ""); +static_assert(ObfuscateId(2) == uint64_t{0x102b90c68120796c}, ""); +static_assert(ObfuscateId(3) == uint64_t{0xd5428022b8669dd5}, ""); +static_assert(ObfuscateId(4) == uint64_t{0x9a596f7eefacc23e}, ""); +static_assert(ObfuscateId(5) == uint64_t{0x5f705edb26f2e6a7}, ""); +static_assert(ObfuscateId(6) == uint64_t{0x24874e375e390b10}, ""); +static_assert(ObfuscateId(7) == uint64_t{0xe99e3d93957f2f79}, ""); +static_assert(ObfuscateId(8) == uint64_t{0xaeb52cefccc553e2}, ""); +static_assert(ObfuscateId(9) == uint64_t{0x73cc1c4c040b784b}, ""); +#ifdef _MSC_VER +#pragma warning(pop) +#endif + +} // namespace + +AudioCodecPairId AudioCodecPairId::Create() { + return AudioCodecPairId(ObfuscateId(GetNextId())); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h new file mode 100644 index 0000000000..b10f14ea66 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h @@ -0,0 +1,74 @@ +/* + * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_ +#define API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_ + +#include <stdint.h> + +#include <utility> + +namespace webrtc { + +class AudioCodecPairId final { + public: + // Copyable, but not default constructible. + AudioCodecPairId() = delete; + AudioCodecPairId(const AudioCodecPairId&) = default; + AudioCodecPairId(AudioCodecPairId&&) = default; + AudioCodecPairId& operator=(const AudioCodecPairId&) = default; + AudioCodecPairId& operator=(AudioCodecPairId&&) = default; + + friend void swap(AudioCodecPairId& a, AudioCodecPairId& b) { + using std::swap; + swap(a.id_, b.id_); + } + + // Creates a new ID, unequal to any previously created ID. + static AudioCodecPairId Create(); + + // IDs can be tested for equality. + friend bool operator==(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ == b.id_; + } + friend bool operator!=(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ != b.id_; + } + + // Comparisons. The ordering of ID values is completely arbitrary, but + // stable, so it's useful e.g. if you want to use IDs as keys in an ordered + // map. + friend bool operator<(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ < b.id_; + } + friend bool operator<=(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ <= b.id_; + } + friend bool operator>=(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ >= b.id_; + } + friend bool operator>(AudioCodecPairId a, AudioCodecPairId b) { + return a.id_ > b.id_; + } + + // Returns a numeric representation of the ID. The numeric values are + // completely arbitrary, but stable, collision-free, and reasonably evenly + // distributed, so they are e.g. useful as hash values in unordered maps. + uint64_t NumericRepresentation() const { return id_; } + + private: + explicit AudioCodecPairId(uint64_t id) : id_(id) {} + + uint64_t id_; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build new file mode 100644 index 0000000000..846946073e --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build @@ -0,0 +1,228 @@ +# 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/audio_codec_pair_id.cc", + "/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc", + "/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc", + "/third_party/libwebrtc/api/audio_codecs/audio_format.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_codecs_api_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc new file mode 100644 index 0000000000..28f5b8aae8 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc @@ -0,0 +1,170 @@ +/* + * Copyright (c) 2012 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/audio_decoder.h" + + +#include <memory> +#include <utility> + +#include "api/array_view.h" +#include "rtc_base/checks.h" +#include "rtc_base/sanitizer.h" +#include "rtc_base/trace_event.h" + +namespace webrtc { + +namespace { + +class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame { + public: + OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload) + : decoder_(decoder), payload_(std::move(payload)) {} + + size_t Duration() const override { + const int ret = decoder_->PacketDuration(payload_.data(), payload_.size()); + return ret < 0 ? 0 : static_cast<size_t>(ret); + } + + absl::optional<DecodeResult> Decode( + rtc::ArrayView<int16_t> decoded) const override { + auto speech_type = AudioDecoder::kSpeech; + const int ret = decoder_->Decode( + payload_.data(), payload_.size(), decoder_->SampleRateHz(), + decoded.size() * sizeof(int16_t), decoded.data(), &speech_type); + return ret < 0 ? absl::nullopt + : absl::optional<DecodeResult>( + {static_cast<size_t>(ret), speech_type}); + } + + private: + AudioDecoder* const decoder_; + const rtc::Buffer payload_; +}; + +} // namespace + +bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const { + return false; +} + +AudioDecoder::ParseResult::ParseResult() = default; +AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default; +AudioDecoder::ParseResult::ParseResult(uint32_t timestamp, + int priority, + std::unique_ptr<EncodedAudioFrame> frame) + : timestamp(timestamp), priority(priority), frame(std::move(frame)) { + RTC_DCHECK_GE(priority, 0); +} + +AudioDecoder::ParseResult::~ParseResult() = default; + +AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=( + ParseResult&& b) = default; + +std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload( + rtc::Buffer&& payload, + uint32_t timestamp) { + std::vector<ParseResult> results; + std::unique_ptr<EncodedAudioFrame> frame( + new OldStyleEncodedFrame(this, std::move(payload))); + results.emplace_back(timestamp, 0, std::move(frame)); + return results; +} + +int AudioDecoder::Decode(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded, + SpeechType* speech_type) { + TRACE_EVENT0("webrtc", "AudioDecoder::Decode"); + rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); + int duration = PacketDuration(encoded, encoded_len); + if (duration >= 0 && + duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { + return -1; + } + return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, + speech_type); +} + +int AudioDecoder::DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded, + SpeechType* speech_type) { + TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant"); + rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len)); + int duration = PacketDurationRedundant(encoded, encoded_len); + if (duration >= 0 && + duration * Channels() * sizeof(int16_t) > max_decoded_bytes) { + return -1; + } + return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded, + speech_type); +} + +int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) { + return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, + speech_type); +} + +bool AudioDecoder::HasDecodePlc() const { + return false; +} + +size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) { + return 0; +} + +// TODO(bugs.webrtc.org/9676): Remove default implementation. +void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/, + rtc::BufferT<int16_t>* /*concealment_audio*/) {} + +int AudioDecoder::ErrorCode() { + return 0; +} + +int AudioDecoder::PacketDuration(const uint8_t* encoded, + size_t encoded_len) const { + return kNotImplemented; +} + +int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const { + return kNotImplemented; +} + +bool AudioDecoder::PacketHasFec(const uint8_t* encoded, + size_t encoded_len) const { + return false; +} + +AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) { + switch (type) { + case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech. + case 1: + return kSpeech; + case 2: + return kComfortNoise; + default: + RTC_DCHECK_NOTREACHED(); + return kSpeech; + } +} + +constexpr int AudioDecoder::kMaxNumberOfChannels; +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder.h new file mode 100644 index 0000000000..41138741bb --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder.h @@ -0,0 +1,195 @@ +/* + * Copyright (c) 2012 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_AUDIO_DECODER_H_ +#define API_AUDIO_CODECS_AUDIO_DECODER_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "rtc_base/buffer.h" + +namespace webrtc { + +class AudioDecoder { + public: + enum SpeechType { + kSpeech = 1, + kComfortNoise = 2, + }; + + // Used by PacketDuration below. Save the value -1 for errors. + enum { kNotImplemented = -2 }; + + AudioDecoder() = default; + virtual ~AudioDecoder() = default; + + AudioDecoder(const AudioDecoder&) = delete; + AudioDecoder& operator=(const AudioDecoder&) = delete; + + class EncodedAudioFrame { + public: + struct DecodeResult { + size_t num_decoded_samples; + SpeechType speech_type; + }; + + virtual ~EncodedAudioFrame() = default; + + // Returns the duration in samples-per-channel of this audio frame. + // If no duration can be ascertained, returns zero. + virtual size_t Duration() const = 0; + + // Returns true if this packet contains DTX. + virtual bool IsDtxPacket() const; + + // Decodes this frame of audio and writes the result in `decoded`. + // `decoded` must be large enough to store as many samples as indicated by a + // call to Duration() . On success, returns an absl::optional containing the + // total number of samples across all channels, as well as whether the + // decoder produced comfort noise or speech. On failure, returns an empty + // absl::optional. Decode may be called at most once per frame object. + virtual absl::optional<DecodeResult> Decode( + rtc::ArrayView<int16_t> decoded) const = 0; + }; + + struct ParseResult { + ParseResult(); + ParseResult(uint32_t timestamp, + int priority, + std::unique_ptr<EncodedAudioFrame> frame); + ParseResult(ParseResult&& b); + ~ParseResult(); + + ParseResult& operator=(ParseResult&& b); + + // The timestamp of the frame is in samples per channel. + uint32_t timestamp; + // The relative priority of the frame compared to other frames of the same + // payload and the same timeframe. A higher value means a lower priority. + // The highest priority is zero - negative values are not allowed. + int priority; + std::unique_ptr<EncodedAudioFrame> frame; + }; + + // Let the decoder parse this payload and prepare zero or more decodable + // frames. Each frame must be between 10 ms and 120 ms long. The caller must + // ensure that the AudioDecoder object outlives any frame objects returned by + // this call. The decoder is free to swap or move the data from the `payload` + // buffer. `timestamp` is the input timestamp, in samples, corresponding to + // the start of the payload. + virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, + uint32_t timestamp); + + // TODO(bugs.webrtc.org/10098): The Decode and DecodeRedundant methods are + // obsolete; callers should call ParsePayload instead. For now, subclasses + // must still implement DecodeInternal. + + // Decodes `encode_len` bytes from `encoded` and writes the result in + // `decoded`. The maximum bytes allowed to be written into `decoded` is + // `max_decoded_bytes`. Returns the total number of samples across all + // channels. If the decoder produced comfort noise, `speech_type` + // is set to kComfortNoise, otherwise it is kSpeech. The desired output + // sample rate is provided in `sample_rate_hz`, which must be valid for the + // codec at hand. + int Decode(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded, + SpeechType* speech_type); + + // Same as Decode(), but interfaces to the decoders redundant decode function. + // The default implementation simply calls the regular Decode() method. + int DecodeRedundant(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + size_t max_decoded_bytes, + int16_t* decoded, + SpeechType* speech_type); + + // Indicates if the decoder implements the DecodePlc method. + virtual bool HasDecodePlc() const; + + // Calls the packet-loss concealment of the decoder to update the state after + // one or several lost packets. The caller has to make sure that the + // memory allocated in `decoded` should accommodate `num_frames` frames. + virtual size_t DecodePlc(size_t num_frames, int16_t* decoded); + + // Asks the decoder to generate packet-loss concealment and append it to the + // end of `concealment_audio`. The concealment audio should be in + // channel-interleaved format, with as many channels as the last decoded + // packet produced. The implementation must produce at least + // requested_samples_per_channel, or nothing at all. This is a signal to the + // caller to conceal the loss with other means. If the implementation provides + // concealment samples, it is also responsible for "stitching" it together + // with the decoded audio on either side of the concealment. + // Note: The default implementation of GeneratePlc will be deleted soon. All + // implementations must provide their own, which can be a simple as a no-op. + // TODO(bugs.webrtc.org/9676): Remove default implementation. + virtual void GeneratePlc(size_t requested_samples_per_channel, + rtc::BufferT<int16_t>* concealment_audio); + + // Resets the decoder state (empty buffers etc.). + virtual void Reset() = 0; + + // Returns the last error code from the decoder. + virtual int ErrorCode(); + + // Returns the duration in samples-per-channel of the payload in `encoded` + // which is `encoded_len` bytes long. Returns kNotImplemented if no duration + // estimate is available, or -1 in case of an error. + virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const; + + // Returns the duration in samples-per-channel of the redandant payload in + // `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no + // duration estimate is available, or -1 in case of an error. + virtual int PacketDurationRedundant(const uint8_t* encoded, + size_t encoded_len) const; + + // Detects whether a packet has forward error correction. The packet is + // comprised of the samples in `encoded` which is `encoded_len` bytes long. + // Returns true if the packet has FEC and false otherwise. + virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const; + + // Returns the actual sample rate of the decoder's output. This value may not + // change during the lifetime of the decoder. + virtual int SampleRateHz() const = 0; + + // The number of channels in the decoder's output. This value may not change + // during the lifetime of the decoder. + virtual size_t Channels() const = 0; + + // The maximum number of audio channels supported by WebRTC decoders. + static constexpr int kMaxNumberOfChannels = 24; + + protected: + static SpeechType ConvertSpeechType(int16_t type); + + virtual int DecodeInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type) = 0; + + virtual int DecodeRedundantInternal(const uint8_t* encoded, + size_t encoded_len, + int sample_rate_hz, + int16_t* decoded, + SpeechType* speech_type); +}; + +} // namespace webrtc +#endif // API_AUDIO_CODECS_AUDIO_DECODER_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h new file mode 100644 index 0000000000..2811f6704b --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h @@ -0,0 +1,53 @@ +/* + * Copyright (c) 2016 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_AUDIO_DECODER_FACTORY_H_ +#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_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 "rtc_base/ref_count.h" + +namespace webrtc { + +// A factory that creates AudioDecoders. +class AudioDecoderFactory : public rtc::RefCountInterface { + public: + virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0; + + virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0; + + // Create a new decoder instance. The `codec_pair_id` argument is used to link + // encoders and decoders that talk to the same remote entity: if a + // AudioEncoderFactory::MakeAudioEncoder() and a + // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that + // compare equal, the factory implementations may assume that the encoder and + // decoder form a pair. (The intended use case for this is to set up + // communication between the AudioEncoder and AudioDecoder instances, which is + // needed for some codecs with built-in bandwidth adaptation.) + // + // Returns null if the format isn't supported. + // + // Note: Implementations need to be robust against combinations other than + // one encoder, one decoder getting the same ID; such decoders must still + // work. + virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id) = 0; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h new file mode 100644 index 0000000000..7ea0c91372 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h @@ -0,0 +1,145 @@ +/* + * 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_AUDIO_DECODER_FACTORY_TEMPLATE_H_ +#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_ + +#include <memory> +#include <vector> + +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +namespace audio_decoder_factory_template_impl { + +template <typename... Ts> +struct Helper; + +// Base case: 0 template parameters. +template <> +struct Helper<> { + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {} + static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; } + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id, + const FieldTrialsView* field_trials) { + return nullptr; + } +}; + +// Inductive case: Called with n + 1 template parameters; calls subroutines +// with n template parameters. +template <typename T, typename... Ts> +struct Helper<T, Ts...> { + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) { + T::AppendSupportedDecoders(specs); + Helper<Ts...>::AppendSupportedDecoders(specs); + } + static bool IsSupportedDecoder(const SdpAudioFormat& format) { + auto opt_config = T::SdpToConfig(format); + static_assert(std::is_same<decltype(opt_config), + absl::optional<typename T::Config>>::value, + "T::SdpToConfig() must return a value of type " + "absl::optional<T::Config>"); + return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format); + } + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id, + const FieldTrialsView* field_trials) { + auto opt_config = T::SdpToConfig(format); + return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id) + : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id, + field_trials); + } +}; + +template <typename... Ts> +class AudioDecoderFactoryT : public AudioDecoderFactory { + public: + explicit AudioDecoderFactoryT(const FieldTrialsView* field_trials) { + field_trials_ = field_trials; + } + + std::vector<AudioCodecSpec> GetSupportedDecoders() override { + std::vector<AudioCodecSpec> specs; + Helper<Ts...>::AppendSupportedDecoders(&specs); + return specs; + } + + bool IsSupportedDecoder(const SdpAudioFormat& format) override { + return Helper<Ts...>::IsSupportedDecoder(format); + } + + std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id) override { + return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id, + field_trials_); + } + + const FieldTrialsView* field_trials_; +}; + +} // namespace audio_decoder_factory_template_impl + +// Make an AudioDecoderFactory that can create instances of the given decoders. +// +// Each decoder type is given as a template argument to the function; it should +// be a struct with the following static member functions: +// +// // Converts `audio_format` to a ConfigType instance. Returns an empty +// // optional if `audio_format` doesn't correctly specify a decoder of our +// // type. +// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format); +// +// // Appends zero or more AudioCodecSpecs to the list that will be returned +// // by AudioDecoderFactory::GetSupportedDecoders(). +// void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs); +// +// // Creates an AudioDecoder for the specified format. Used to implement +// // AudioDecoderFactory::MakeAudioDecoder(). +// std::unique_ptr<AudioDecoder> MakeAudioDecoder( +// const ConfigType& config, +// absl::optional<AudioCodecPairId> codec_pair_id); +// +// ConfigType should be a type that encapsulates all the settings needed to +// create an AudioDecoder. T::Config (where T is the decoder struct) should +// either be the config type, or an alias for it. +// +// Whenever it tries to do something, the new factory will try each of the +// decoder types in the order they were specified in the template argument +// list, stopping at the first one that claims to be able to do the job. +// +// TODO(kwiberg): Point at CreateBuiltinAudioDecoderFactory() for an example of +// how it is used. +template <typename... Ts> +rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory( + const FieldTrialsView* field_trials = nullptr) { + // There's no technical reason we couldn't allow zero template parameters, + // but such a factory couldn't create any decoders, and callers can do this + // by mistake by simply forgetting the <> altogether. So we forbid it in + // order to prevent caller foot-shooting. + static_assert(sizeof...(Ts) >= 1, + "Caller must give at least one template parameter"); + + return rtc::make_ref_counted< + audio_decoder_factory_template_impl::AudioDecoderFactoryT<Ts...>>( + field_trials); +} + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc new file mode 100644 index 0000000000..31bb8739f7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc @@ -0,0 +1,114 @@ +/* + * Copyright (c) 2014 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/audio_encoder.h" + +#include "rtc_base/checks.h" +#include "rtc_base/trace_event.h" + +namespace webrtc { + +ANAStats::ANAStats() = default; +ANAStats::~ANAStats() = default; +ANAStats::ANAStats(const ANAStats&) = default; + +AudioEncoder::EncodedInfo::EncodedInfo() = default; +AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default; +AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default; +AudioEncoder::EncodedInfo::~EncodedInfo() = default; +AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=( + const EncodedInfo&) = default; +AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) = + default; + +int AudioEncoder::RtpTimestampRateHz() const { + return SampleRateHz(); +} + +AudioEncoder::EncodedInfo AudioEncoder::Encode( + uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) { + TRACE_EVENT0("webrtc", "AudioEncoder::Encode"); + RTC_CHECK_EQ(audio.size(), + static_cast<size_t>(NumChannels() * SampleRateHz() / 100)); + + const size_t old_size = encoded->size(); + EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded); + RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes); + return info; +} + +bool AudioEncoder::SetFec(bool enable) { + return !enable; +} + +bool AudioEncoder::SetDtx(bool enable) { + return !enable; +} + +bool AudioEncoder::GetDtx() const { + return false; +} + +bool AudioEncoder::SetApplication(Application application) { + return false; +} + +void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {} + +void AudioEncoder::SetTargetBitrate(int target_bps) {} + +rtc::ArrayView<std::unique_ptr<AudioEncoder>> +AudioEncoder::ReclaimContainedEncoders() { + return nullptr; +} + +bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string, + RtcEventLog* event_log) { + return false; +} + +void AudioEncoder::DisableAudioNetworkAdaptor() {} + +void AudioEncoder::OnReceivedUplinkPacketLossFraction( + float uplink_packet_loss_fraction) {} + +void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction( + float uplink_recoverable_packet_loss_fraction) { + RTC_DCHECK_NOTREACHED(); +} + +void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) { + OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt); +} + +void AudioEncoder::OnReceivedUplinkBandwidth( + int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms) {} + +void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) { + OnReceivedUplinkBandwidth(update.target_bitrate.bps(), + update.bwe_period.ms()); +} + +void AudioEncoder::OnReceivedRtt(int rtt_ms) {} + +void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {} + +void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms, + int max_frame_length_ms) {} + +ANAStats AudioEncoder::GetANAStats() const { + return ANAStats(); +} + +constexpr int AudioEncoder::kMaxNumberOfChannels; +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h new file mode 100644 index 0000000000..7f5a34214f --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h @@ -0,0 +1,260 @@ +/* + * Copyright (c) 2014 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_AUDIO_ENCODER_H_ +#define API_AUDIO_CODECS_AUDIO_ENCODER_H_ + +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/base/attributes.h" +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "api/call/bitrate_allocation.h" +#include "api/units/time_delta.h" +#include "rtc_base/buffer.h" + +namespace webrtc { + +class RtcEventLog; + +// Statistics related to Audio Network Adaptation. +struct ANAStats { + ANAStats(); + ANAStats(const ANAStats&); + ~ANAStats(); + // Number of actions taken by the ANA bitrate controller since the start of + // the call. If this value is not set, it indicates that the bitrate + // controller is disabled. + absl::optional<uint32_t> bitrate_action_counter; + // Number of actions taken by the ANA channel controller since the start of + // the call. If this value is not set, it indicates that the channel + // controller is disabled. + absl::optional<uint32_t> channel_action_counter; + // Number of actions taken by the ANA DTX controller since the start of the + // call. If this value is not set, it indicates that the DTX controller is + // disabled. + absl::optional<uint32_t> dtx_action_counter; + // Number of actions taken by the ANA FEC controller since the start of the + // call. If this value is not set, it indicates that the FEC controller is + // disabled. + absl::optional<uint32_t> fec_action_counter; + // Number of times the ANA frame length controller decided to increase the + // frame length since the start of the call. If this value is not set, it + // indicates that the frame length controller is disabled. + absl::optional<uint32_t> frame_length_increase_counter; + // Number of times the ANA frame length controller decided to decrease the + // frame length since the start of the call. If this value is not set, it + // indicates that the frame length controller is disabled. + absl::optional<uint32_t> frame_length_decrease_counter; + // The uplink packet loss fractions as set by the ANA FEC controller. If this + // value is not set, it indicates that the ANA FEC controller is not active. + absl::optional<float> uplink_packet_loss_fraction; +}; + +// This is the interface class for encoders in AudioCoding module. Each codec +// type must have an implementation of this class. +class AudioEncoder { + public: + // Used for UMA logging of codec usage. The same codecs, with the + // same values, must be listed in + // src/tools/metrics/histograms/histograms.xml in chromium to log + // correct values. + enum class CodecType { + kOther = 0, // Codec not specified, and/or not listed in this enum + kOpus = 1, + kIsac = 2, + kPcmA = 3, + kPcmU = 4, + kG722 = 5, + kIlbc = 6, + + // Number of histogram bins in the UMA logging of codec types. The + // total number of different codecs that are logged cannot exceed this + // number. + kMaxLoggedAudioCodecTypes + }; + + struct EncodedInfoLeaf { + size_t encoded_bytes = 0; + uint32_t encoded_timestamp = 0; + int payload_type = 0; + bool send_even_if_empty = false; + bool speech = true; + CodecType encoder_type = CodecType::kOther; + }; + + // This is the main struct for auxiliary encoding information. Each encoded + // packet should be accompanied by one EncodedInfo struct, containing the + // total number of `encoded_bytes`, the `encoded_timestamp` and the + // `payload_type`. If the packet contains redundant encodings, the `redundant` + // vector will be populated with EncodedInfoLeaf structs. Each struct in the + // vector represents one encoding; the order of structs in the vector is the + // same as the order in which the actual payloads are written to the byte + // stream. When EncoderInfoLeaf structs are present in the vector, the main + // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the + // vector. + struct EncodedInfo : public EncodedInfoLeaf { + EncodedInfo(); + EncodedInfo(const EncodedInfo&); + EncodedInfo(EncodedInfo&&); + ~EncodedInfo(); + EncodedInfo& operator=(const EncodedInfo&); + EncodedInfo& operator=(EncodedInfo&&); + + std::vector<EncodedInfoLeaf> redundant; + }; + + virtual ~AudioEncoder() = default; + + // Returns the input sample rate in Hz and the number of input channels. + // These are constants set at instantiation time. + virtual int SampleRateHz() const = 0; + virtual size_t NumChannels() const = 0; + + // Returns the rate at which the RTP timestamps are updated. The default + // implementation returns SampleRateHz(). + virtual int RtpTimestampRateHz() const; + + // Returns the number of 10 ms frames the encoder will put in the next + // packet. This value may only change when Encode() outputs a packet; i.e., + // the encoder may vary the number of 10 ms frames from packet to packet, but + // it must decide the length of the next packet no later than when outputting + // the preceding packet. + virtual size_t Num10MsFramesInNextPacket() const = 0; + + // Returns the maximum value that can be returned by + // Num10MsFramesInNextPacket(). + virtual size_t Max10MsFramesInAPacket() const = 0; + + // Returns the current target bitrate in bits/s. The value -1 means that the + // codec adapts the target automatically, and a current target cannot be + // provided. + virtual int GetTargetBitrate() const = 0; + + // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 * + // NumChannels() samples). Multi-channel audio must be sample-interleaved. + // The encoder appends zero or more bytes of output to `encoded` and returns + // additional encoding information. Encode() checks some preconditions, calls + // EncodeImpl() which does the actual work, and then checks some + // postconditions. + EncodedInfo Encode(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded); + + // Resets the encoder to its starting state, discarding any input that has + // been fed to the encoder but not yet emitted in a packet. + virtual void Reset() = 0; + + // Enables or disables codec-internal FEC (forward error correction). Returns + // true if the codec was able to comply. The default implementation returns + // true when asked to disable FEC and false when asked to enable it (meaning + // that FEC isn't supported). + virtual bool SetFec(bool enable); + + // Enables or disables codec-internal VAD/DTX. Returns true if the codec was + // able to comply. The default implementation returns true when asked to + // disable DTX and false when asked to enable it (meaning that DTX isn't + // supported). + virtual bool SetDtx(bool enable); + + // Returns the status of codec-internal DTX. The default implementation always + // returns false. + virtual bool GetDtx() const; + + // Sets the application mode. Returns true if the codec was able to comply. + // The default implementation just returns false. + enum class Application { kSpeech, kAudio }; + virtual bool SetApplication(Application application); + + // Tells the encoder about the highest sample rate the decoder is expected to + // use when decoding the bitstream. The encoder would typically use this + // information to adjust the quality of the encoding. The default + // implementation does nothing. + virtual void SetMaxPlaybackRate(int frequency_hz); + + // Tells the encoder what average bitrate we'd like it to produce. The + // encoder is free to adjust or disregard the given bitrate (the default + // implementation does the latter). + ABSL_DEPRECATED("Use OnReceivedTargetAudioBitrate instead") + virtual void SetTargetBitrate(int target_bps); + + // Causes this encoder to let go of any other encoders it contains, and + // returns a pointer to an array where they are stored (which is required to + // live as long as this encoder). Unless the returned array is empty, you may + // not call any methods on this encoder afterwards, except for the + // destructor. The default implementation just returns an empty array. + // NOTE: This method is subject to change. Do not call or override it. + virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>> + ReclaimContainedEncoders(); + + // Enables audio network adaptor. Returns true if successful. + virtual bool EnableAudioNetworkAdaptor(const std::string& config_string, + RtcEventLog* event_log); + + // Disables audio network adaptor. + virtual void DisableAudioNetworkAdaptor(); + + // Provides uplink packet loss fraction to this encoder to allow it to adapt. + // `uplink_packet_loss_fraction` is in the range [0.0, 1.0]. + virtual void OnReceivedUplinkPacketLossFraction( + float uplink_packet_loss_fraction); + + ABSL_DEPRECATED("") + virtual void OnReceivedUplinkRecoverablePacketLossFraction( + float uplink_recoverable_packet_loss_fraction); + + // Provides target audio bitrate to this encoder to allow it to adapt. + virtual void OnReceivedTargetAudioBitrate(int target_bps); + + // Provides target audio bitrate and corresponding probing interval of + // the bandwidth estimator to this encoder to allow it to adapt. + virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps, + absl::optional<int64_t> bwe_period_ms); + + // Provides target audio bitrate and corresponding probing interval of + // the bandwidth estimator to this encoder to allow it to adapt. + virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update); + + // Provides RTT to this encoder to allow it to adapt. + virtual void OnReceivedRtt(int rtt_ms); + + // Provides overhead to this encoder to adapt. The overhead is the number of + // bytes that will be added to each packet the encoder generates. + virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet); + + // To allow encoder to adapt its frame length, it must be provided the frame + // length range that receivers can accept. + virtual void SetReceiverFrameLengthRange(int min_frame_length_ms, + int max_frame_length_ms); + + // Get statistics related to audio network adaptation. + virtual ANAStats GetANAStats() const; + + // The range of frame lengths that are supported or nullopt if there's no sch + // information. This is used to calculated the full bitrate range, including + // overhead. + virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange() + const = 0; + + // The maximum number of audio channels supported by WebRTC encoders. + static constexpr int kMaxNumberOfChannels = 24; + + protected: + // Subclasses implement this to perform the actual encoding. Called by + // Encode(). + virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp, + rtc::ArrayView<const int16_t> audio, + rtc::Buffer* encoded) = 0; +}; +} // namespace webrtc +#endif // API_AUDIO_CODECS_AUDIO_ENCODER_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h new file mode 100644 index 0000000000..6128b1b6f3 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h @@ -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. + */ + +#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_ +#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_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 "rtc_base/ref_count.h" + +namespace webrtc { + +// A factory that creates AudioEncoders. +class AudioEncoderFactory : public rtc::RefCountInterface { + public: + // Returns a prioritized list of audio codecs, to use for signaling etc. + virtual std::vector<AudioCodecSpec> GetSupportedEncoders() = 0; + + // Returns information about how this format would be encoded, provided it's + // supported. More format and format variations may be supported than those + // returned by GetSupportedEncoders(). + virtual absl::optional<AudioCodecInfo> QueryAudioEncoder( + const SdpAudioFormat& format) = 0; + + // Creates an AudioEncoder for the specified format. The encoder will tags its + // payloads with the specified payload type. The `codec_pair_id` argument is + // used to link encoders and decoders that talk to the same remote entity: if + // a AudioEncoderFactory::MakeAudioEncoder() and a + // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that + // compare equal, the factory implementations may assume that the encoder and + // decoder form a pair. (The intended use case for this is to set up + // communication between the AudioEncoder and AudioDecoder instances, which is + // needed for some codecs with built-in bandwidth adaptation.) + // + // Returns null if the format isn't supported. + // + // Note: Implementations need to be robust against combinations other than + // one encoder, one decoder getting the same ID; such encoders must still + // work. + // + // TODO(ossu): Try to avoid audio encoders having to know their payload type. + virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder( + int payload_type, + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id) = 0; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h new file mode 100644 index 0000000000..8a70ba2268 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h @@ -0,0 +1,163 @@ +/* + * 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_AUDIO_ENCODER_FACTORY_TEMPLATE_H_ +#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_ + +#include <memory> +#include <vector> + +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/field_trials_view.h" +#include "api/make_ref_counted.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +namespace audio_encoder_factory_template_impl { + +template <typename... Ts> +struct Helper; + +// Base case: 0 template parameters. +template <> +struct Helper<> { + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {} + static absl::optional<AudioCodecInfo> QueryAudioEncoder( + const SdpAudioFormat& format) { + return absl::nullopt; + } + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + int payload_type, + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id, + const FieldTrialsView* field_trials) { + return nullptr; + } +}; + +// Inductive case: Called with n + 1 template parameters; calls subroutines +// with n template parameters. +template <typename T, typename... Ts> +struct Helper<T, Ts...> { + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) { + T::AppendSupportedEncoders(specs); + Helper<Ts...>::AppendSupportedEncoders(specs); + } + static absl::optional<AudioCodecInfo> QueryAudioEncoder( + const SdpAudioFormat& format) { + auto opt_config = T::SdpToConfig(format); + static_assert(std::is_same<decltype(opt_config), + absl::optional<typename T::Config>>::value, + "T::SdpToConfig() must return a value of type " + "absl::optional<T::Config>"); + return opt_config ? absl::optional<AudioCodecInfo>( + T::QueryAudioEncoder(*opt_config)) + : Helper<Ts...>::QueryAudioEncoder(format); + } + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + int payload_type, + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id, + const FieldTrialsView* field_trials) { + auto opt_config = T::SdpToConfig(format); + if (opt_config) { + return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id); + } else { + return Helper<Ts...>::MakeAudioEncoder(payload_type, format, + codec_pair_id, field_trials); + } + } +}; + +template <typename... Ts> +class AudioEncoderFactoryT : public AudioEncoderFactory { + public: + explicit AudioEncoderFactoryT(const FieldTrialsView* field_trials) { + field_trials_ = field_trials; + } + + std::vector<AudioCodecSpec> GetSupportedEncoders() override { + std::vector<AudioCodecSpec> specs; + Helper<Ts...>::AppendSupportedEncoders(&specs); + return specs; + } + + absl::optional<AudioCodecInfo> QueryAudioEncoder( + const SdpAudioFormat& format) override { + return Helper<Ts...>::QueryAudioEncoder(format); + } + + std::unique_ptr<AudioEncoder> MakeAudioEncoder( + int payload_type, + const SdpAudioFormat& format, + absl::optional<AudioCodecPairId> codec_pair_id) override { + return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id, + field_trials_); + } + + const FieldTrialsView* field_trials_; +}; + +} // namespace audio_encoder_factory_template_impl + +// Make an AudioEncoderFactory that can create instances of the given encoders. +// +// Each encoder type is given as a template argument to the function; it should +// be a struct with the following static member functions: +// +// // Converts `audio_format` to a ConfigType instance. Returns an empty +// // optional if `audio_format` doesn't correctly specify an encoder of our +// // type. +// absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format); +// +// // Appends zero or more AudioCodecSpecs to the list that will be returned +// // by AudioEncoderFactory::GetSupportedEncoders(). +// void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); +// +// // Returns information about how this format would be encoded. Used to +// // implement AudioEncoderFactory::QueryAudioEncoder(). +// AudioCodecInfo QueryAudioEncoder(const ConfigType& config); +// +// // Creates an AudioEncoder for the specified format. Used to implement +// // AudioEncoderFactory::MakeAudioEncoder(). +// std::unique_ptr<AudioDecoder> MakeAudioEncoder( +// const ConfigType& config, +// int payload_type, +// absl::optional<AudioCodecPairId> codec_pair_id); +// +// ConfigType should be a type that encapsulates all the settings needed to +// create an AudioEncoder. T::Config (where T is the encoder struct) should +// either be the config type, or an alias for it. +// +// Whenever it tries to do something, the new factory will try each of the +// encoders in the order they were specified in the template argument list, +// stopping at the first one that claims to be able to do the job. +// +// TODO(kwiberg): Point at CreateBuiltinAudioEncoderFactory() for an example of +// how it is used. +template <typename... Ts> +rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory( + const FieldTrialsView* field_trials = nullptr) { + // There's no technical reason we couldn't allow zero template parameters, + // but such a factory couldn't create any encoders, and callers can do this + // by mistake by simply forgetting the <> altogether. So we forbid it in + // order to prevent caller foot-shooting. + static_assert(sizeof...(Ts) >= 1, + "Caller must give at least one template parameter"); + + return rtc::make_ref_counted< + audio_encoder_factory_template_impl::AudioEncoderFactoryT<Ts...>>( + field_trials); +} + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.cc b/third_party/libwebrtc/api/audio_codecs/audio_format.cc new file mode 100644 index 0000000000..2a529a49ee --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_format.cc @@ -0,0 +1,86 @@ +/* + * Copyright (c) 2016 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/audio_format.h" + +#include <utility> + +#include "absl/strings/match.h" + +namespace webrtc { + +SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default; +SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default; + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels) + : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {} + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + const Parameters& param) + : name(name), + clockrate_hz(clockrate_hz), + num_channels(num_channels), + parameters(param) {} + +SdpAudioFormat::SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + Parameters&& param) + : name(name), + clockrate_hz(clockrate_hz), + num_channels(num_channels), + parameters(std::move(param)) {} + +bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const { + return absl::EqualsIgnoreCase(name, o.name) && + clockrate_hz == o.clockrate_hz && num_channels == o.num_channels; +} + +SdpAudioFormat::~SdpAudioFormat() = default; +SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default; +SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default; + +bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) { + return absl::EqualsIgnoreCase(a.name, b.name) && + a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels && + a.parameters == b.parameters; +} + +AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, + size_t num_channels, + int bitrate_bps) + : AudioCodecInfo(sample_rate_hz, + num_channels, + bitrate_bps, + bitrate_bps, + bitrate_bps) {} + +AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, + size_t num_channels, + int default_bitrate_bps, + int min_bitrate_bps, + int max_bitrate_bps) + : sample_rate_hz(sample_rate_hz), + num_channels(num_channels), + default_bitrate_bps(default_bitrate_bps), + min_bitrate_bps(min_bitrate_bps), + max_bitrate_bps(max_bitrate_bps) { + RTC_DCHECK_GT(sample_rate_hz, 0); + RTC_DCHECK_GT(num_channels, 0); + RTC_DCHECK_GE(min_bitrate_bps, 0); + RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); + RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.h b/third_party/libwebrtc/api/audio_codecs/audio_format.h new file mode 100644 index 0000000000..0cf67799b8 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/audio_format.h @@ -0,0 +1,133 @@ +/* + * Copyright (c) 2016 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_AUDIO_FORMAT_H_ +#define API_AUDIO_CODECS_AUDIO_FORMAT_H_ + +#include <stddef.h> + +#include <map> +#include <string> + +#include "absl/strings/string_view.h" +#include "rtc_base/checks.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// SDP specification for a single audio codec. +struct RTC_EXPORT SdpAudioFormat { + using Parameters = std::map<std::string, std::string>; + + SdpAudioFormat(const SdpAudioFormat&); + SdpAudioFormat(SdpAudioFormat&&); + SdpAudioFormat(absl::string_view name, int clockrate_hz, size_t num_channels); + SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + const Parameters& param); + SdpAudioFormat(absl::string_view name, + int clockrate_hz, + size_t num_channels, + Parameters&& param); + ~SdpAudioFormat(); + + // Returns true if this format is compatible with `o`. In SDP terminology: + // would it represent the same codec between an offer and an answer? As + // opposed to operator==, this method disregards codec parameters. + bool Matches(const SdpAudioFormat& o) const; + + SdpAudioFormat& operator=(const SdpAudioFormat&); + SdpAudioFormat& operator=(SdpAudioFormat&&); + + friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b); + friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) { + return !(a == b); + } + + std::string name; + int clockrate_hz; + size_t num_channels; + Parameters parameters; +}; + +// Information about how an audio format is treated by the codec implementation. +// Contains basic information, such as sample rate and number of channels, which +// isn't uniformly presented by SDP. Also contains flags indicating support for +// integrating with other parts of WebRTC, like external VAD and comfort noise +// level calculation. +// +// To avoid API breakage, and make the code clearer, AudioCodecInfo should not +// be directly initializable with any flags indicating optional support. If it +// were, these initializers would break any time a new flag was added. It's also +// more difficult to understand: +// AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true}; +// than +// AudioCodecInfo info(16000, 1, 32000); +// info.allow_comfort_noise = true; +// info.future_flag_b = true; +// info.future_flag_c = true; +struct AudioCodecInfo { + AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps); + AudioCodecInfo(int sample_rate_hz, + size_t num_channels, + int default_bitrate_bps, + int min_bitrate_bps, + int max_bitrate_bps); + AudioCodecInfo(const AudioCodecInfo& b) = default; + ~AudioCodecInfo() = default; + + bool operator==(const AudioCodecInfo& b) const { + return sample_rate_hz == b.sample_rate_hz && + num_channels == b.num_channels && + default_bitrate_bps == b.default_bitrate_bps && + min_bitrate_bps == b.min_bitrate_bps && + max_bitrate_bps == b.max_bitrate_bps && + allow_comfort_noise == b.allow_comfort_noise && + supports_network_adaption == b.supports_network_adaption; + } + + bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); } + + bool HasFixedBitrate() const { + RTC_DCHECK_GE(min_bitrate_bps, 0); + RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps); + RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps); + return min_bitrate_bps == max_bitrate_bps; + } + + int sample_rate_hz; + size_t num_channels; + int default_bitrate_bps; + int min_bitrate_bps; + int max_bitrate_bps; + + bool allow_comfort_noise = true; // This codec can be used with an external + // comfort noise generator. + bool supports_network_adaption = false; // This codec can adapt to varying + // network conditions. +}; + +// AudioCodecSpec ties an audio format to specific information about the codec +// and its implementation. +struct AudioCodecSpec { + bool operator==(const AudioCodecSpec& b) const { + return format == b.format && info == b.info; + } + + bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); } + + SdpAudioFormat format; + AudioCodecInfo info; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_AUDIO_FORMAT_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc new file mode 100644 index 0000000000..881113d985 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc @@ -0,0 +1,68 @@ +/* + * 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/builtin_audio_decoder_factory.h" + +#include <memory> +#include <vector> + +#include "api/audio_codecs/L16/audio_decoder_L16.h" +#include "api/audio_codecs/audio_decoder_factory_template.h" +#include "api/audio_codecs/g711/audio_decoder_g711.h" +#include "api/audio_codecs/g722/audio_decoder_g722.h" +#if WEBRTC_USE_BUILTIN_ILBC +#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck +#endif +#if WEBRTC_USE_BUILTIN_OPUS +#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" +#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck +#endif + +namespace webrtc { + +namespace { + +// Modify an audio decoder to not advertise support for anything. +template <typename T> +struct NotAdvertised { + using Config = typename T::Config; + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + return T::SdpToConfig(audio_format); + } + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) { + // Don't advertise support for anything. + } + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const Config& config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) { + return T::MakeAudioDecoder(config, codec_pair_id); + } +}; + +} // namespace + +rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() { + return CreateAudioDecoderFactory< + +#if WEBRTC_USE_BUILTIN_OPUS + AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>, +#endif + + AudioDecoderG722, + +#if WEBRTC_USE_BUILTIN_ILBC + AudioDecoderIlbc, +#endif + + AudioDecoderG711, NotAdvertised<AudioDecoderL16>>(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h new file mode 100644 index 0000000000..72e1e3d96e --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h @@ -0,0 +1,28 @@ +/* + * Copyright (c) 2016 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_BUILTIN_AUDIO_DECODER_FACTORY_H_ +#define API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ + +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// Creates a new factory that can create the built-in types of audio decoders. +// Note: This will link with all the code implementing those codecs, so if you +// only need a subset of the codecs, consider using +// CreateAudioDecoderFactory<...codecs listed here...>() or +// CreateOpusAudioDecoderFactory() instead. +rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory(); + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build new file mode 100644 index 0000000000..366307ea13 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build @@ -0,0 +1,234 @@ +# 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" +DEFINES["WEBRTC_USE_BUILTIN_ILBC"] = "1" +DEFINES["WEBRTC_USE_BUILTIN_OPUS"] = "1" + +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/builtin_audio_decoder_factory.cc" +] + +if not CONFIG["MOZ_DEBUG"]: + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0" + DEFINES["NDEBUG"] = True + DEFINES["NVALGRIND"] = True + +if CONFIG["MOZ_DEBUG"] == "1": + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1" + +if CONFIG["OS_TARGET"] == "Android": + + DEFINES["ANDROID"] = True + DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1" + DEFINES["HAVE_SYS_UIO_H"] = True + DEFINES["WEBRTC_ANDROID"] = True + DEFINES["WEBRTC_ANDROID_OPENSLES"] = True + DEFINES["WEBRTC_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": + + 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("builtin_audio_decoder_factory_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.cc new file mode 100644 index 0000000000..4546a2eaee --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.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/builtin_audio_encoder_factory.h" + +#include <memory> +#include <vector> + +#include "api/audio_codecs/L16/audio_encoder_L16.h" +#include "api/audio_codecs/audio_encoder_factory_template.h" +#include "api/audio_codecs/g711/audio_encoder_g711.h" +#include "api/audio_codecs/g722/audio_encoder_g722.h" +#if WEBRTC_USE_BUILTIN_ILBC +#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck +#endif +#if WEBRTC_USE_BUILTIN_OPUS +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" +#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck +#endif + +namespace webrtc { + +namespace { + +// Modify an audio encoder to not advertise support for anything. +template <typename T> +struct NotAdvertised { + using Config = typename T::Config; + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + return T::SdpToConfig(audio_format); + } + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) { + // Don't advertise support for anything. + } + static AudioCodecInfo QueryAudioEncoder(const Config& config) { + return T::QueryAudioEncoder(config); + } + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr) { + return T::MakeAudioEncoder(config, payload_type, codec_pair_id, + field_trials); + } +}; + +} // namespace + +rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() { + return CreateAudioEncoderFactory< + +#if WEBRTC_USE_BUILTIN_OPUS + AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>, +#endif + + AudioEncoderG722, + +#if WEBRTC_USE_BUILTIN_ILBC + AudioEncoderIlbc, +#endif + + AudioEncoderG711, NotAdvertised<AudioEncoderL16>>(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h new file mode 100644 index 0000000000..f833de10f1 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h @@ -0,0 +1,28 @@ +/* + * Copyright (c) 2016 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_BUILTIN_AUDIO_ENCODER_FACTORY_H_ +#define API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ + +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// Creates a new factory that can create the built-in types of audio encoders. +// Note: This will link with all the code implementing those codecs, so if you +// only need a subset of the codecs, consider using +// CreateAudioEncoderFactory<...codecs listed here...>() or +// CreateOpusAudioEncoderFactory() instead. +rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory(); + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build new file mode 100644 index 0000000000..db0e3fbe00 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build @@ -0,0 +1,234 @@ +# 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" +DEFINES["WEBRTC_USE_BUILTIN_ILBC"] = "1" +DEFINES["WEBRTC_USE_BUILTIN_OPUS"] = "1" + +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/builtin_audio_encoder_factory.cc" +] + +if not CONFIG["MOZ_DEBUG"]: + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0" + DEFINES["NDEBUG"] = True + DEFINES["NVALGRIND"] = True + +if CONFIG["MOZ_DEBUG"] == "1": + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1" + +if CONFIG["OS_TARGET"] == "Android": + + DEFINES["ANDROID"] = True + DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1" + DEFINES["HAVE_SYS_UIO_H"] = True + DEFINES["WEBRTC_ANDROID"] = True + DEFINES["WEBRTC_ANDROID_OPENSLES"] = True + DEFINES["WEBRTC_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": + + 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("builtin_audio_encoder_factory_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn new file mode 100644 index 0000000000..b2ff324f12 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn @@ -0,0 +1,55 @@ +# 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_library("audio_encoder_g711") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_encoder_g711.cc", + "audio_encoder_g711.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:g711", + "../../../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_g711") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_g711.cc", + "audio_decoder_g711.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:g711", + "../../../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/g711/audio_decoder_g711.cc b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc new file mode 100644 index 0000000000..838f7e9624 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc @@ -0,0 +1,67 @@ +/* + * 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/g711/audio_decoder_g711.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" +#include "rtc_base/numerics/safe_conversions.h" + +namespace webrtc { + +absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig( + const SdpAudioFormat& format) { + const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU"); + const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA"); + if (format.clockrate_hz == 8000 && format.num_channels >= 1 && + (is_pcmu || is_pcma)) { + Config config; + config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA; + config.num_channels = rtc::dchecked_cast<int>(format.num_channels); + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; + } else { + return absl::nullopt; + } +} + +void AudioDecoderG711::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + for (const char* type : {"PCMU", "PCMA"}) { + specs->push_back({{type, 8000, 1}, {8000, 1, 64000}}); + } +} + +std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder( + const Config& config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + switch (config.type) { + case Config::Type::kPcmU: + return std::make_unique<AudioDecoderPcmU>(config.num_channels); + case Config::Type::kPcmA: + return std::make_unique<AudioDecoderPcmA>(config.num_channels); + default: + RTC_DCHECK_NOTREACHED(); + return nullptr; + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h new file mode 100644 index 0000000000..0f7a98d345 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h @@ -0,0 +1,49 @@ +/* + * 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_G711_AUDIO_DECODER_G711_H_ +#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_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 { + +// G711 decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct RTC_EXPORT AudioDecoderG711 { + struct Config { + enum class Type { kPcmU, kPcmA }; + bool IsOk() const { + return (type == Type::kPcmU || type == Type::kPcmA) && + num_channels >= 1 && + num_channels <= AudioDecoder::kMaxNumberOfChannels; + } + Type type; + 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( + const Config& config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build new file mode 100644 index 0000000000..4782d01dd1 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_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/g711/audio_decoder_g711.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_g711_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc new file mode 100644 index 0000000000..1dca3b80d3 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc @@ -0,0 +1,95 @@ +/* + * 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/g711/audio_encoder_g711.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.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<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig( + const SdpAudioFormat& format) { + const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU"); + const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA"); + if (format.clockrate_hz == 8000 && format.num_channels >= 1 && + (is_pcmu || is_pcma)) { + Config config; + config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA; + config.num_channels = rtc::dchecked_cast<int>(format.num_channels); + config.frame_size_ms = 20; + auto ptime_iter = format.parameters.find("ptime"); + if (ptime_iter != format.parameters.end()) { + const auto ptime = rtc::StringToNumber<int>(ptime_iter->second); + if (ptime && *ptime > 0) { + config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60); + } + } + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; + } else { + return absl::nullopt; + } +} + +void AudioEncoderG711::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + for (const char* type : {"PCMU", "PCMA"}) { + specs->push_back({{type, 8000, 1}, {8000, 1, 64000}}); + } +} + +AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) { + RTC_DCHECK(config.IsOk()); + return {8000, rtc::dchecked_cast<size_t>(config.num_channels), + 64000 * config.num_channels}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + switch (config.type) { + case Config::Type::kPcmU: { + AudioEncoderPcmU::Config impl_config; + impl_config.num_channels = config.num_channels; + impl_config.frame_size_ms = config.frame_size_ms; + impl_config.payload_type = payload_type; + return std::make_unique<AudioEncoderPcmU>(impl_config); + } + case Config::Type::kPcmA: { + AudioEncoderPcmA::Config impl_config; + impl_config.num_channels = config.num_channels; + impl_config.frame_size_ms = config.frame_size_ms; + impl_config.payload_type = payload_type; + return std::make_unique<AudioEncoderPcmA>(impl_config); + } + default: { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h new file mode 100644 index 0000000000..4b3eb845e0 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h @@ -0,0 +1,54 @@ +/* + * 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_G711_AUDIO_ENCODER_G711_H_ +#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_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/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// G711 encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderG711 { + struct Config { + enum class Type { kPcmU, kPcmA }; + bool IsOk() const { + return (type == Type::kPcmU || type == Type::kPcmA) && + frame_size_ms > 0 && frame_size_ms % 10 == 0 && + num_channels >= 1 && + num_channels <= AudioEncoder::kMaxNumberOfChannels; + } + Type type = Type::kPcmU; + int num_channels = 1; + int frame_size_ms = 20; + }; + static absl::optional<AudioEncoderG711::Config> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build new file mode 100644 index 0000000000..c972978c13 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_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/g711/audio_encoder_g711.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_g711_gn") 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") diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn new file mode 100644 index 0000000000..22cf48220f --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn @@ -0,0 +1,58 @@ +# 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_ilbc_config") { + visibility = [ "*" ] + sources = [ "audio_encoder_ilbc_config.h" ] +} + +rtc_library("audio_encoder_ilbc") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_encoder_ilbc.cc", + "audio_encoder_ilbc.h", + ] + deps = [ + ":audio_encoder_ilbc_config", + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:ilbc", + "../../../rtc_base:safe_conversions", + "../../../rtc_base:safe_minmax", + "../../../rtc_base:stringutils", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("audio_decoder_ilbc") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_ilbc.cc", + "audio_decoder_ilbc.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:ilbc", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc new file mode 100644 index 0000000000..c58316903a --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc @@ -0,0 +1,42 @@ +/* + * 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/ilbc/audio_decoder_ilbc.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" + +namespace webrtc { + +absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig( + const SdpAudioFormat& format) { + if (absl::EqualsIgnoreCase(format.name, "ILBC") && + format.clockrate_hz == 8000 && format.num_channels == 1) { + return Config(); + } + return absl::nullopt; +} + +void AudioDecoderIlbc::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}}); +} + +std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder( + Config config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + return std::make_unique<AudioDecoderIlbcImpl>(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h new file mode 100644 index 0000000000..60566c88df --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h @@ -0,0 +1,39 @@ +/* + * 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_ILBC_AUDIO_DECODER_ILBC_H_ +#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_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" + +namespace webrtc { + +// ILBC decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct AudioDecoderIlbc { + struct Config {}; // Empty---no config values needed! + 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_ILBC_AUDIO_DECODER_ILBC_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build new file mode 100644 index 0000000000..53e9d1a4a7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build @@ -0,0 +1,232 @@ +# 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/ilbc/audio_decoder_ilbc.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 + + 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": + + 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_ilbc_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc new file mode 100644 index 0000000000..b497948491 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc @@ -0,0 +1,88 @@ +/* + * 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/ilbc/audio_encoder_ilbc.h" + +#include <memory> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" +#include "rtc_base/string_to_number.h" + +namespace webrtc { +namespace { +int GetIlbcBitrate(int ptime) { + switch (ptime) { + case 20: + case 40: + // 38 bytes per frame of 20 ms => 15200 bits/s. + return 15200; + case 30: + case 60: + // 50 bytes per frame of 30 ms => (approx) 13333 bits/s. + return 13333; + default: + RTC_CHECK_NOTREACHED(); + } +} +} // namespace + +absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig( + const SdpAudioFormat& format) { + if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") || + format.clockrate_hz != 8000 || format.num_channels != 1) { + return absl::nullopt; + } + + AudioEncoderIlbcConfig config; + 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, 20, 60); + } + } + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; +} + +void AudioEncoderIlbc::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + const SdpAudioFormat fmt = {"ILBC", 8000, 1}; + const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); + specs->push_back({fmt, info}); +} + +AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder( + const AudioEncoderIlbcConfig& config) { + RTC_DCHECK(config.IsOk()); + return {8000, 1, GetIlbcBitrate(config.frame_size_ms)}; +} + +std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder( + const AudioEncoderIlbcConfig& 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<AudioEncoderIlbcImpl>(config, payload_type); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h new file mode 100644 index 0000000000..a5306841ce --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.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_ILBC_AUDIO_ENCODER_ILBC_H_ +#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_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/ilbc/audio_encoder_ilbc_config.h" +#include "api/field_trials_view.h" + +namespace webrtc { + +// ILBC encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct AudioEncoderIlbc { + using Config = AudioEncoderIlbcConfig; + static absl::optional<AudioEncoderIlbcConfig> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const AudioEncoderIlbcConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h new file mode 100644 index 0000000000..4d82f9901c --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h @@ -0,0 +1,28 @@ +/* + * 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_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_ +#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_ + +namespace webrtc { + +struct AudioEncoderIlbcConfig { + bool IsOk() const { + return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 || + frame_size_ms == 60); + } + int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms. + // Note that frame size 40 ms produces encodings with two 20 ms frames in + // them, and frame size 60 ms consists of two 30 ms frames. +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_gn/moz.build new file mode 100644 index 0000000000..75737b8f19 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_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("audio_encoder_ilbc_config_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build new file mode 100644 index 0000000000..bddfe42193 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build @@ -0,0 +1,232 @@ +# 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/ilbc/audio_encoder_ilbc.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 + + 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": + + 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_ilbc_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn new file mode 100644 index 0000000000..eb90a0b9ac --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn @@ -0,0 +1,110 @@ +# 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_library("audio_encoder_opus_config") { + visibility = [ "*" ] + sources = [ + "audio_encoder_multi_channel_opus_config.cc", + "audio_encoder_multi_channel_opus_config.h", + "audio_encoder_opus_config.cc", + "audio_encoder_opus_config.h", + ] + deps = [ "../../../rtc_base/system:rtc_export" ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] + defines = [] + if (rtc_opus_variable_complexity) { + defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ] + } else { + defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ] + } +} + +rtc_source_set("audio_decoder_opus_config") { + visibility = [ "*" ] + sources = [ "audio_decoder_multi_channel_opus_config.h" ] + deps = [ "..:audio_codecs_api" ] +} + +rtc_library("audio_encoder_opus") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + public = [ "audio_encoder_opus.h" ] + sources = [ "audio_encoder_opus.cc" ] + deps = [ + ":audio_encoder_opus_config", + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("audio_decoder_opus") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_opus.cc", + "audio_decoder_opus.h", + ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_opus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} + +rtc_library("audio_encoder_multiopus") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + public = [ "audio_encoder_multi_channel_opus.h" ] + sources = [ "audio_encoder_multi_channel_opus.cc" ] + deps = [ + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_multiopus", + "../../../rtc_base/system:rtc_export", + "../opus:audio_encoder_opus_config", + ] + absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] +} + +rtc_library("audio_decoder_multiopus") { + visibility = [ "*" ] + poisonous = [ "audio_codecs" ] + sources = [ + "audio_decoder_multi_channel_opus.cc", + "audio_decoder_multi_channel_opus.h", + ] + deps = [ + ":audio_decoder_opus_config", + "..:audio_codecs_api", + "../../../api:field_trials_view", + "../../../modules/audio_coding:webrtc_multiopus", + "../../../rtc_base/system:rtc_export", + ] + absl_deps = [ + "//third_party/abseil-cpp/absl/memory", + "//third_party/abseil-cpp/absl/strings", + "//third_party/abseil-cpp/absl/types:optional", + ] +} diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc new file mode 100644 index 0000000000..0fb4e05511 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc @@ -0,0 +1,71 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" + +#include <memory> +#include <utility> +#include <vector> + +#include "absl/memory/memory.h" +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h" + +namespace webrtc { + +absl::optional<AudioDecoderMultiChannelOpusConfig> +AudioDecoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) { + return AudioDecoderMultiChannelOpusImpl::SdpToConfig(format); +} + +void AudioDecoderMultiChannelOpus::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + // To get full utilization of the surround support of the Opus lib, we can + // mark which channel is the low frequency effects (LFE). But that is not done + // ATM. + { + AudioCodecInfo surround_5_1_opus_info{48000, 6, + /* default_bitrate_bps= */ 128000}; + surround_5_1_opus_info.allow_comfort_noise = false; + surround_5_1_opus_info.supports_network_adaption = false; + SdpAudioFormat opus_format({"multiopus", + 48000, + 6, + {{"minptime", "10"}, + {"useinbandfec", "1"}, + {"channel_mapping", "0,4,1,2,3,5"}, + {"num_streams", "4"}, + {"coupled_streams", "2"}}}); + specs->push_back({std::move(opus_format), surround_5_1_opus_info}); + } + { + AudioCodecInfo surround_7_1_opus_info{48000, 8, + /* default_bitrate_bps= */ 200000}; + surround_7_1_opus_info.allow_comfort_noise = false; + surround_7_1_opus_info.supports_network_adaption = false; + SdpAudioFormat opus_format({"multiopus", + 48000, + 8, + {{"minptime", "10"}, + {"useinbandfec", "1"}, + {"channel_mapping", "0,6,1,2,3,4,5,7"}, + {"num_streams", "5"}, + {"coupled_streams", "3"}}}); + specs->push_back({std::move(opus_format), surround_7_1_opus_info}); + } +} + +std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder( + AudioDecoderMultiChannelOpusConfig config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config); +} +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h new file mode 100644 index 0000000000..eafd6c6939 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_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/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Opus decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct RTC_EXPORT AudioDecoderMultiChannelOpus { + using Config = AudioDecoderMultiChannelOpusConfig; + static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs); + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + AudioDecoderMultiChannelOpusConfig config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h new file mode 100644 index 0000000000..f97c5c3193 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_ + +#include <vector> + +#include "api/audio_codecs/audio_decoder.h" + +namespace webrtc { +struct AudioDecoderMultiChannelOpusConfig { + // The number of channels that the decoder will output. + int num_channels; + + // Number of mono or stereo encoded Opus streams. + int num_streams; + + // Number of channel pairs coupled together, see RFC 7845 section + // 5.1.1. Has to be less than the number of streams. + int coupled_streams; + + // Channel mapping table, defines the mapping from encoded streams to output + // channels. See RFC 7845 section 5.1.1. + std::vector<unsigned char> channel_mapping; + + bool IsOk() const { + if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels || + num_streams < 0 || coupled_streams < 0) { + return false; + } + if (num_streams < coupled_streams) { + return false; + } + if (channel_mapping.size() != static_cast<size_t>(num_channels)) { + return false; + } + + // Every mono stream codes one channel, every coupled stream codes two. This + // is the total coded channel count: + const int max_coded_channel = num_streams + coupled_streams; + for (const auto& x : channel_mapping) { + // Coded channels >= max_coded_channel don't exist. Except for 255, which + // tells Opus to put silence in output channel x. + if (x >= max_coded_channel && x != 255) { + return false; + } + } + + if (num_channels > 255 || max_coded_channel >= 255) { + return false; + } + return true; + } +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build new file mode 100644 index 0000000000..2b2bc6d9a7 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build @@ -0,0 +1,226 @@ +# 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", + "/media/libopus/include/", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.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_multiopus_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc new file mode 100644 index 0000000000..efc9a73546 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc @@ -0,0 +1,86 @@ +/* + * 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/opus/audio_decoder_opus.h" + +#include <memory> +#include <utility> +#include <vector> + +#include "absl/strings/match.h" +#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" + +namespace webrtc { + +bool AudioDecoderOpus::Config::IsOk() const { + if (sample_rate_hz != 16000 && sample_rate_hz != 48000) { + // Unsupported sample rate. (libopus supports a few other rates as + // well; we can add support for them when needed.) + return false; + } + if (num_channels != 1 && num_channels != 2) { + return false; + } + return true; +} + +absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig( + const SdpAudioFormat& format) { + const auto num_channels = [&]() -> absl::optional<int> { + auto stereo = format.parameters.find("stereo"); + if (stereo != format.parameters.end()) { + if (stereo->second == "0") { + return 1; + } else if (stereo->second == "1") { + return 2; + } else { + return absl::nullopt; // Bad stereo parameter. + } + } + return 1; // Default to mono. + }(); + if (absl::EqualsIgnoreCase(format.name, "opus") && + format.clockrate_hz == 48000 && format.num_channels == 2 && + num_channels) { + Config config; + config.num_channels = *num_channels; + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return absl::nullopt; + } + return config; + } else { + return absl::nullopt; + } +} + +void AudioDecoderOpus::AppendSupportedDecoders( + std::vector<AudioCodecSpec>* specs) { + AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000}; + opus_info.allow_comfort_noise = false; + opus_info.supports_network_adaption = true; + SdpAudioFormat opus_format( + {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}); + specs->push_back({std::move(opus_format), opus_info}); +} + +std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder( + Config config, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return std::make_unique<AudioDecoderOpusImpl>(config.num_channels, + config.sample_rate_hz); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.h new file mode 100644 index 0000000000..138c0377df --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.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_OPUS_AUDIO_DECODER_OPUS_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_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 { + +// Opus decoder API for use as a template parameter to +// CreateAudioDecoderFactory<...>(). +struct RTC_EXPORT AudioDecoderOpus { + struct Config { + bool IsOk() const; // Checks if the values are currently OK. + int sample_rate_hz = 48000; + int num_channels = 1; + }; + 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_OPUS_AUDIO_DECODER_OPUS_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_config_gn/moz.build new file mode 100644 index 0000000000..e2c470d5ee --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_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_decoder_opus_config_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build new file mode 100644 index 0000000000..58e6355a55 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build @@ -0,0 +1,233 @@ +# This Source Code Form is subject to the terms of the Mozilla Public +# License, v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. + + + ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ### + ### DO NOT edit it by hand. ### + +COMPILE_FLAGS["OS_INCLUDES"] = [] +AllowCompilerWarnings() + +DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1" +DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True +DEFINES["RTC_ENABLE_VP9"] = True +DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0" +DEFINES["WEBRTC_LIBRARY_IMPL"] = True +DEFINES["WEBRTC_MOZILLA_BUILD"] = True +DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0" +DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0" + +FINAL_LIBRARY = "webrtc" + + +LOCAL_INCLUDES += [ + "!/ipc/ipdl/_ipdlheaders", + "!/third_party/libwebrtc/gen", + "/ipc/chromium/src", + "/media/libopus/include/", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.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 + + 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": + + 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_opus_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc new file mode 100644 index 0000000000..14f480b1ec --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc @@ -0,0 +1,75 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" + +#include <utility> + +#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h" + +namespace webrtc { + +absl::optional<AudioEncoderMultiChannelOpusConfig> +AudioEncoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) { + return AudioEncoderMultiChannelOpusImpl::SdpToConfig(format); +} + +void AudioEncoderMultiChannelOpus::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + // To get full utilization of the surround support of the Opus lib, we can + // mark which channel is the low frequency effects (LFE). But that is not done + // ATM. + { + AudioCodecInfo surround_5_1_opus_info{48000, 6, + /* default_bitrate_bps= */ 128000}; + surround_5_1_opus_info.allow_comfort_noise = false; + surround_5_1_opus_info.supports_network_adaption = false; + SdpAudioFormat opus_format({"multiopus", + 48000, + 6, + {{"minptime", "10"}, + {"useinbandfec", "1"}, + {"channel_mapping", "0,4,1,2,3,5"}, + {"num_streams", "4"}, + {"coupled_streams", "2"}}}); + specs->push_back({std::move(opus_format), surround_5_1_opus_info}); + } + { + AudioCodecInfo surround_7_1_opus_info{48000, 8, + /* default_bitrate_bps= */ 200000}; + surround_7_1_opus_info.allow_comfort_noise = false; + surround_7_1_opus_info.supports_network_adaption = false; + SdpAudioFormat opus_format({"multiopus", + 48000, + 8, + {{"minptime", "10"}, + {"useinbandfec", "1"}, + {"channel_mapping", "0,6,1,2,3,4,5,7"}, + {"num_streams", "5"}, + {"coupled_streams", "3"}}}); + specs->push_back({std::move(opus_format), surround_7_1_opus_info}); + } +} + +AudioCodecInfo AudioEncoderMultiChannelOpus::QueryAudioEncoder( + const AudioEncoderMultiChannelOpusConfig& config) { + return AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(config); +} + +std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder( + const AudioEncoderMultiChannelOpusConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config, + payload_type); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h new file mode 100644 index 0000000000..c1c4db3577 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h @@ -0,0 +1,43 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_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/opus/audio_encoder_multi_channel_opus_config.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Opus encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderMultiChannelOpus { + using Config = AudioEncoderMultiChannelOpusConfig; + static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const Config& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc new file mode 100644 index 0000000000..0052c429b2 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc @@ -0,0 +1,106 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h" + +namespace webrtc { + +namespace { +constexpr int kDefaultComplexity = 9; +} // namespace + +AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig() + : frame_size_ms(kDefaultFrameSizeMs), + num_channels(1), + application(ApplicationMode::kVoip), + bitrate_bps(32000), + fec_enabled(false), + cbr_enabled(false), + dtx_enabled(false), + max_playback_rate_hz(48000), + complexity(kDefaultComplexity), + num_streams(-1), + coupled_streams(-1) {} +AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig( + const AudioEncoderMultiChannelOpusConfig&) = default; +AudioEncoderMultiChannelOpusConfig::~AudioEncoderMultiChannelOpusConfig() = + default; +AudioEncoderMultiChannelOpusConfig& AudioEncoderMultiChannelOpusConfig:: +operator=(const AudioEncoderMultiChannelOpusConfig&) = default; + +bool AudioEncoderMultiChannelOpusConfig::IsOk() const { + if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) + return false; + if (num_channels >= 255) { + return false; + } + if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps) + return false; + if (complexity < 0 || complexity > 10) + return false; + + // Check the lengths: + if (num_streams < 0 || coupled_streams < 0) { + return false; + } + if (num_streams < coupled_streams) { + return false; + } + if (channel_mapping.size() != static_cast<size_t>(num_channels)) { + return false; + } + + // Every mono stream codes one channel, every coupled stream codes two. This + // is the total coded channel count: + const int max_coded_channel = num_streams + coupled_streams; + for (const auto& x : channel_mapping) { + // Coded channels >= max_coded_channel don't exist. Except for 255, which + // tells Opus to ignore input channel x. + if (x >= max_coded_channel && x != 255) { + return false; + } + } + + // Inverse mapping. + constexpr int kNotSet = -1; + std::vector<int> coded_channels_to_input_channels(max_coded_channel, kNotSet); + for (size_t i = 0; i < num_channels; ++i) { + if (channel_mapping[i] == 255) { + continue; + } + + // If it's not ignored, put it in the inverted mapping. But first check if + // we've told Opus to use another input channel for this coded channel: + const int coded_channel = channel_mapping[i]; + if (coded_channels_to_input_channels[coded_channel] != kNotSet) { + // Coded channel `coded_channel` comes from both input channels + // `coded_channels_to_input_channels[coded_channel]` and `i`. + return false; + } + + coded_channels_to_input_channels[coded_channel] = i; + } + + // Check that we specified what input the encoder should use to produce + // every coded channel. + for (int i = 0; i < max_coded_channel; ++i) { + if (coded_channels_to_input_channels[i] == kNotSet) { + // Coded channel `i` has unspecified input channel. + return false; + } + } + + if (num_channels > 255 || max_coded_channel >= 255) { + return false; + } + return true; +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h new file mode 100644 index 0000000000..9b51246c15 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_ + +#include <stddef.h> + +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio_codecs/opus/audio_encoder_opus_config.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +struct RTC_EXPORT AudioEncoderMultiChannelOpusConfig { + static constexpr int kDefaultFrameSizeMs = 20; + + // Opus API allows a min bitrate of 500bps, but Opus documentation suggests + // bitrate should be in the range of 6000 to 510000, inclusive. + static constexpr int kMinBitrateBps = 6000; + static constexpr int kMaxBitrateBps = 510000; + + AudioEncoderMultiChannelOpusConfig(); + AudioEncoderMultiChannelOpusConfig(const AudioEncoderMultiChannelOpusConfig&); + ~AudioEncoderMultiChannelOpusConfig(); + AudioEncoderMultiChannelOpusConfig& operator=( + const AudioEncoderMultiChannelOpusConfig&); + + int frame_size_ms; + size_t num_channels; + enum class ApplicationMode { kVoip, kAudio }; + ApplicationMode application; + int bitrate_bps; + bool fec_enabled; + bool cbr_enabled; + bool dtx_enabled; + int max_playback_rate_hz; + std::vector<int> supported_frame_lengths_ms; + + int complexity; + + // Number of mono/stereo Opus streams. + int num_streams; + + // Number of channel pairs coupled together, see RFC 7845 section + // 5.1.1. Has to be less than the number of streams + int coupled_streams; + + // Channel mapping table, defines the mapping from encoded streams to input + // channels. See RFC 7845 section 5.1.1. + std::vector<unsigned char> channel_mapping; + + bool IsOk() const; +}; + +} // namespace webrtc +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build new file mode 100644 index 0000000000..91afd0a4e4 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build @@ -0,0 +1,226 @@ +# 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", + "/media/libopus/include/", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.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_multiopus_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc new file mode 100644 index 0000000000..5b6322da4c --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc @@ -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. + */ + +#include "api/audio_codecs/opus/audio_encoder_opus.h" + +#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h" + +namespace webrtc { + +absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig( + const SdpAudioFormat& format) { + return AudioEncoderOpusImpl::SdpToConfig(format); +} + +void AudioEncoderOpus::AppendSupportedEncoders( + std::vector<AudioCodecSpec>* specs) { + AudioEncoderOpusImpl::AppendSupportedEncoders(specs); +} + +AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder( + const AudioEncoderOpusConfig& config) { + return AudioEncoderOpusImpl::QueryAudioEncoder(config); +} + +std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder( + const AudioEncoderOpusConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/, + const FieldTrialsView* field_trials) { + if (!config.IsOk()) { + RTC_DCHECK_NOTREACHED(); + return nullptr; + } + return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.h new file mode 100644 index 0000000000..df93ae5303 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.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_OPUS_AUDIO_ENCODER_OPUS_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_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/opus/audio_encoder_opus_config.h" +#include "api/field_trials_view.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +// Opus encoder API for use as a template parameter to +// CreateAudioEncoderFactory<...>(). +struct RTC_EXPORT AudioEncoderOpus { + using Config = AudioEncoderOpusConfig; + static absl::optional<AudioEncoderOpusConfig> SdpToConfig( + const SdpAudioFormat& audio_format); + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs); + static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config); + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const AudioEncoderOpusConfig& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr); +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc new file mode 100644 index 0000000000..a9ab924b38 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc @@ -0,0 +1,75 @@ +/* + * 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/opus/audio_encoder_opus_config.h" + +namespace webrtc { + +namespace { + +#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) +constexpr int kDefaultComplexity = 5; +#else +constexpr int kDefaultComplexity = 9; +#endif + +constexpr int kDefaultLowRateComplexity = + WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity; + +} // namespace + +constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs; +constexpr int AudioEncoderOpusConfig::kMinBitrateBps; +constexpr int AudioEncoderOpusConfig::kMaxBitrateBps; + +AudioEncoderOpusConfig::AudioEncoderOpusConfig() + : frame_size_ms(kDefaultFrameSizeMs), + sample_rate_hz(48000), + num_channels(1), + application(ApplicationMode::kVoip), + bitrate_bps(32000), + fec_enabled(false), + cbr_enabled(false), + max_playback_rate_hz(48000), + complexity(kDefaultComplexity), + low_rate_complexity(kDefaultLowRateComplexity), + complexity_threshold_bps(12500), + complexity_threshold_window_bps(1500), + dtx_enabled(false), + uplink_bandwidth_update_interval_ms(200), + payload_type(-1) {} +AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) = + default; +AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default; +AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=( + const AudioEncoderOpusConfig&) = default; + +bool AudioEncoderOpusConfig::IsOk() const { + if (frame_size_ms <= 0 || frame_size_ms % 10 != 0) + return false; + if (sample_rate_hz != 16000 && sample_rate_hz != 48000) { + // Unsupported input sample rate. (libopus supports a few other rates as + // well; we can add support for them when needed.) + return false; + } + if (num_channels >= 255) { + return false; + } + if (!bitrate_bps) + return false; + if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps) + return false; + if (complexity < 0 || complexity > 10) + return false; + if (low_rate_complexity < 0 || low_rate_complexity > 10) + return false; + return true; +} +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h new file mode 100644 index 0000000000..d5d7256c70 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h @@ -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. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ + +#include <stddef.h> + +#include <vector> + +#include "absl/types/optional.h" +#include "rtc_base/system/rtc_export.h" + +namespace webrtc { + +struct RTC_EXPORT AudioEncoderOpusConfig { + static constexpr int kDefaultFrameSizeMs = 20; + + // Opus API allows a min bitrate of 500bps, but Opus documentation suggests + // bitrate should be in the range of 6000 to 510000, inclusive. + static constexpr int kMinBitrateBps = 6000; + static constexpr int kMaxBitrateBps = 510000; + + AudioEncoderOpusConfig(); + AudioEncoderOpusConfig(const AudioEncoderOpusConfig&); + ~AudioEncoderOpusConfig(); + AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&); + + bool IsOk() const; // Checks if the values are currently OK. + + int frame_size_ms; + int sample_rate_hz; + size_t num_channels; + enum class ApplicationMode { kVoip, kAudio }; + ApplicationMode application; + + // NOTE: This member must always be set. + // TODO(kwiberg): Turn it into just an int. + absl::optional<int> bitrate_bps; + + bool fec_enabled; + bool cbr_enabled; + int max_playback_rate_hz; + + // `complexity` is used when the bitrate goes above + // `complexity_threshold_bps` + `complexity_threshold_window_bps`; + // `low_rate_complexity` is used when the bitrate falls below + // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the + // interval in the middle, we keep using the most recent of the two + // complexity settings. + int complexity; + int low_rate_complexity; + int complexity_threshold_bps; + int complexity_threshold_window_bps; + + bool dtx_enabled; + std::vector<int> supported_frame_lengths_ms; + int uplink_bandwidth_update_interval_ms; + + // NOTE: This member isn't necessary, and will soon go away. See + // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847 + int payload_type; +}; + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build new file mode 100644 index 0000000000..06732b48f4 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build @@ -0,0 +1,222 @@ +# 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_OPUS_VARIABLE_COMPLEXITY"] = "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" +] + +SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc" +] + +if not CONFIG["MOZ_DEBUG"]: + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0" + DEFINES["NDEBUG"] = True + DEFINES["NVALGRIND"] = True + +if CONFIG["MOZ_DEBUG"] == "1": + + DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1" + +if CONFIG["OS_TARGET"] == "Android": + + DEFINES["ANDROID"] = True + DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1" + DEFINES["HAVE_SYS_UIO_H"] = True + DEFINES["WEBRTC_ANDROID"] = True + DEFINES["WEBRTC_ANDROID_OPENSLES"] = True + DEFINES["WEBRTC_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": + + 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_opus_config_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build new file mode 100644 index 0000000000..ab84d3f755 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build @@ -0,0 +1,233 @@ +# This Source Code Form is subject to the terms of the Mozilla Public +# License, v. 2.0. If a copy of the MPL was not distributed with this +# file, You can obtain one at http://mozilla.org/MPL/2.0/. + + + ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ### + ### DO NOT edit it by hand. ### + +COMPILE_FLAGS["OS_INCLUDES"] = [] +AllowCompilerWarnings() + +DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1" +DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True +DEFINES["RTC_ENABLE_VP9"] = True +DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0" +DEFINES["WEBRTC_LIBRARY_IMPL"] = True +DEFINES["WEBRTC_MOZILLA_BUILD"] = True +DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0" +DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0" + +FINAL_LIBRARY = "webrtc" + + +LOCAL_INCLUDES += [ + "!/ipc/ipdl/_ipdlheaders", + "!/third_party/libwebrtc/gen", + "/ipc/chromium/src", + "/media/libopus/include/", + "/third_party/libwebrtc/", + "/third_party/libwebrtc/third_party/abseil-cpp/", + "/tools/profiler/public" +] + +UNIFIED_SOURCES += [ + "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.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 + + 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": + + 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_opus_gn") diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc new file mode 100644 index 0000000000..ed68f2584e --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc @@ -0,0 +1,49 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/opus_audio_decoder_factory.h" + +#include <memory> +#include <vector> + +#include "api/audio_codecs/audio_decoder_factory_template.h" +#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" +#include "api/audio_codecs/opus/audio_decoder_opus.h" + +namespace webrtc { + +namespace { + +// Modify an audio decoder to not advertise support for anything. +template <typename T> +struct NotAdvertised { + using Config = typename T::Config; + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + return T::SdpToConfig(audio_format); + } + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) { + // Don't advertise support for anything. + } + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const Config& config, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) { + return T::MakeAudioDecoder(config, codec_pair_id); + } +}; + +} // namespace + +rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory() { + return CreateAudioDecoderFactory< + AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>>(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h new file mode 100644 index 0000000000..b4f497f8ff --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h @@ -0,0 +1,26 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_ + +#include "api/audio_codecs/audio_decoder_factory.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// Creates a new factory that can create only Opus audio decoders. Works like +// CreateAudioDecoderFactory<AudioDecoderOpus>(), but is easier to use and is +// not inline because it isn't a template. +rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory(); + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc new file mode 100644 index 0000000000..8c286f21e1 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "api/audio_codecs/opus_audio_encoder_factory.h" + +#include <memory> +#include <vector> + +#include "api/audio_codecs/audio_encoder_factory_template.h" +#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" +#include "api/audio_codecs/opus/audio_encoder_opus.h" + +namespace webrtc { +namespace { + +// Modify an audio encoder to not advertise support for anything. +template <typename T> +struct NotAdvertised { + using Config = typename T::Config; + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + return T::SdpToConfig(audio_format); + } + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) { + // Don't advertise support for anything. + } + static AudioCodecInfo QueryAudioEncoder(const Config& config) { + return T::QueryAudioEncoder(config); + } + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config& config, + int payload_type, + absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt, + const FieldTrialsView* field_trials = nullptr) { + return T::MakeAudioEncoder(config, payload_type, codec_pair_id, + field_trials); + } +}; + +} // namespace + +rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory() { + return CreateAudioEncoderFactory< + AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>>(); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h new file mode 100644 index 0000000000..8c1683b6f5 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h @@ -0,0 +1,26 @@ +/* + * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_ +#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_ + +#include "api/audio_codecs/audio_encoder_factory.h" +#include "api/scoped_refptr.h" + +namespace webrtc { + +// Creates a new factory that can create only Opus audio encoders. Works like +// CreateAudioEncoderFactory<AudioEncoderOpus>(), but is easier to use and is +// not inline because it isn't a template. +rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory(); + +} // namespace webrtc + +#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_ diff --git a/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn new file mode 100644 index 0000000000..89f5fef1ea --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn @@ -0,0 +1,39 @@ +# 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") +} + +if (rtc_include_tests) { + rtc_library("audio_codecs_api_unittests") { + testonly = true + sources = [ + "audio_decoder_factory_template_unittest.cc", + "audio_encoder_factory_template_unittest.cc", + ] + deps = [ + "..:audio_codecs_api", + "../../../test:audio_codec_mocks", + "../../../test:scoped_key_value_config", + "../../../test:test_support", + "../L16:audio_decoder_L16", + "../L16:audio_encoder_L16", + "../g711:audio_decoder_g711", + "../g711:audio_encoder_g711", + "../g722:audio_decoder_g722", + "../g722:audio_encoder_g722", + "../ilbc:audio_decoder_ilbc", + "../ilbc:audio_encoder_ilbc", + "../opus:audio_decoder_opus", + "../opus:audio_encoder_opus", + ] + } +} diff --git a/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc new file mode 100644 index 0000000000..0b18cf934a --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc @@ -0,0 +1,222 @@ +/* + * 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/audio_decoder_factory_template.h" + +#include <memory> + +#include "api/audio_codecs/L16/audio_decoder_L16.h" +#include "api/audio_codecs/g711/audio_decoder_g711.h" +#include "api/audio_codecs/g722/audio_decoder_g722.h" +#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" +#include "api/audio_codecs/opus/audio_decoder_opus.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/mock_audio_decoder.h" +#include "test/scoped_key_value_config.h" + +namespace webrtc { + +namespace { + +struct BogusParams { + static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; } + static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; } +}; + +struct ShamParams { + static SdpAudioFormat AudioFormat() { + return {"sham", 16000, 2, {{"param", "value"}}}; + } + static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; } +}; + +template <typename Params> +struct AudioDecoderFakeApi { + struct Config { + SdpAudioFormat audio_format; + }; + + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + if (Params::AudioFormat() == audio_format) { + Config config = {audio_format}; + return config; + } else { + return absl::nullopt; + } + } + + static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) { + specs->push_back({Params::AudioFormat(), Params::CodecInfo()}); + } + + static AudioCodecInfo QueryAudioDecoder(const Config&) { + return Params::CodecInfo(); + } + + static std::unique_ptr<AudioDecoder> MakeAudioDecoder( + const Config&, + absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) { + auto dec = std::make_unique<testing::StrictMock<MockAudioDecoder>>(); + EXPECT_CALL(*dec, SampleRateHz()) + .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz)); + EXPECT_CALL(*dec, Die()); + return std::move(dec); + } +}; + +} // namespace + +TEST(AudioDecoderFactoryTemplateTest, NoDecoderTypes) { + test::ScopedKeyValueConfig field_trials; + rtc::scoped_refptr<AudioDecoderFactory> factory( + rtc::make_ref_counted< + audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>( + &field_trials)); + EXPECT_THAT(factory->GetSupportedDecoders(), ::testing::IsEmpty()); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt)); +} + +TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) { + auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>>(); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt)); + auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec); + EXPECT_EQ(8000, dec->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTemplateTest, TwoDecoderTypes) { + auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>, + AudioDecoderFakeApi<ShamParams>>(); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}, + AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}}, + {16000, 2, 23456}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1})); + EXPECT_TRUE( + factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt)); + auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec1); + EXPECT_EQ(8000, dec1->SampleRateHz()); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"sham", 16000, 2}, absl::nullopt)); + auto dec2 = factory->MakeAudioDecoder( + {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt); + ASSERT_NE(nullptr, dec2); + EXPECT_EQ(16000, dec2->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTemplateTest, G711) { + auto factory = CreateAudioDecoderFactory<AudioDecoderG711>(); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}}, + AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"pcmu", 16000, 1}, absl::nullopt)); + auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec1); + EXPECT_EQ(8000, dec1->SampleRateHz()); + auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec2); + EXPECT_EQ(8000, dec2->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTemplateTest, G722) { + auto factory = CreateAudioDecoderFactory<AudioDecoderG722>(); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt)); + auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec1); + EXPECT_EQ(16000, dec1->SampleRateHz()); + EXPECT_EQ(1u, dec1->Channels()); + auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, absl::nullopt); + ASSERT_NE(nullptr, dec2); + EXPECT_EQ(16000, dec2->SampleRateHz()); + EXPECT_EQ(2u, dec2->Channels()); + auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, absl::nullopt); + ASSERT_EQ(nullptr, dec3); +} + +TEST(AudioDecoderFactoryTemplateTest, Ilbc) { + auto factory = CreateAudioDecoderFactory<AudioDecoderIlbc>(); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 8000, 1}, absl::nullopt)); + auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, dec); + EXPECT_EQ(8000, dec->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTemplateTest, L16) { + auto factory = CreateAudioDecoderFactory<AudioDecoderL16>(); + EXPECT_THAT( + factory->GetSupportedDecoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}}, + AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}}, + AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}}, + AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}}, + AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}}, + AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}})); + EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1})); + EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"L16", 8000, 0}, absl::nullopt)); + auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, absl::nullopt); + ASSERT_NE(nullptr, dec); + EXPECT_EQ(48000, dec->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTemplateTest, Opus) { + auto factory = CreateAudioDecoderFactory<AudioDecoderOpus>(); + AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000}; + opus_info.allow_comfort_noise = false; + opus_info.supports_network_adaption = true; + const SdpAudioFormat opus_format( + {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}); + EXPECT_THAT(factory->GetSupportedDecoders(), + ::testing::ElementsAre(AudioCodecSpec{opus_format, opus_info})); + EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1})); + EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2})); + EXPECT_EQ(nullptr, + factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt)); + auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, absl::nullopt); + ASSERT_NE(nullptr, dec); + EXPECT_EQ(48000, dec->SampleRateHz()); +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc new file mode 100644 index 0000000000..dbba387724 --- /dev/null +++ b/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc @@ -0,0 +1,224 @@ +/* + * 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/audio_encoder_factory_template.h" + +#include <memory> + +#include "api/audio_codecs/L16/audio_encoder_L16.h" +#include "api/audio_codecs/g711/audio_encoder_g711.h" +#include "api/audio_codecs/g722/audio_encoder_g722.h" +#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" +#include "api/audio_codecs/opus/audio_encoder_opus.h" +#include "test/gmock.h" +#include "test/gtest.h" +#include "test/mock_audio_encoder.h" +#include "test/scoped_key_value_config.h" + +namespace webrtc { + +namespace { + +struct BogusParams { + static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; } + static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; } +}; + +struct ShamParams { + static SdpAudioFormat AudioFormat() { + return {"sham", 16000, 2, {{"param", "value"}}}; + } + static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; } +}; + +template <typename Params> +struct AudioEncoderFakeApi { + struct Config { + SdpAudioFormat audio_format; + }; + + static absl::optional<Config> SdpToConfig( + const SdpAudioFormat& audio_format) { + if (Params::AudioFormat() == audio_format) { + Config config = {audio_format}; + return config; + } else { + return absl::nullopt; + } + } + + static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) { + specs->push_back({Params::AudioFormat(), Params::CodecInfo()}); + } + + static AudioCodecInfo QueryAudioEncoder(const Config&) { + return Params::CodecInfo(); + } + + static std::unique_ptr<AudioEncoder> MakeAudioEncoder( + const Config&, + int payload_type, + absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) { + auto enc = std::make_unique<testing::StrictMock<MockAudioEncoder>>(); + EXPECT_CALL(*enc, SampleRateHz()) + .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz)); + return std::move(enc); + } +}; + +} // namespace + +TEST(AudioEncoderFactoryTemplateTest, NoEncoderTypes) { + test::ScopedKeyValueConfig field_trials; + rtc::scoped_refptr<AudioEncoderFactory> factory( + rtc::make_ref_counted< + audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>( + &field_trials)); + EXPECT_THAT(factory->GetSupportedEncoders(), ::testing::IsEmpty()); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt)); +} + +TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) { + auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>>(); + EXPECT_THAT(factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ(AudioCodecInfo(8000, 1, 12345), + factory->QueryAudioEncoder({"bogus", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt)); + auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc); + EXPECT_EQ(8000, enc->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, TwoEncoderTypes) { + auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>, + AudioEncoderFakeApi<ShamParams>>(); + EXPECT_THAT(factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}, + AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}}, + {16000, 2, 23456}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ(AudioCodecInfo(8000, 1, 12345), + factory->QueryAudioEncoder({"bogus", 8000, 1})); + EXPECT_EQ( + AudioCodecInfo(16000, 2, 23456), + factory->QueryAudioEncoder({"sham", 16000, 2, {{"param", "value"}}})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt)); + auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc1); + EXPECT_EQ(8000, enc1->SampleRateHz()); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"sham", 16000, 2}, absl::nullopt)); + auto enc2 = factory->MakeAudioEncoder( + 17, {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt); + ASSERT_NE(nullptr, enc2); + EXPECT_EQ(16000, enc2->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, G711) { + auto factory = CreateAudioEncoderFactory<AudioEncoderG711>(); + EXPECT_THAT(factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}}, + AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1})); + EXPECT_EQ(AudioCodecInfo(8000, 1, 64000), + factory->QueryAudioEncoder({"PCMA", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, absl::nullopt)); + auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc1); + EXPECT_EQ(8000, enc1->SampleRateHz()); + auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc2); + EXPECT_EQ(8000, enc2->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, G722) { + auto factory = CreateAudioEncoderFactory<AudioEncoderG722>(); + EXPECT_THAT(factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ(AudioCodecInfo(16000, 1, 64000), + factory->QueryAudioEncoder({"G722", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt)); + auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc); + EXPECT_EQ(16000, enc->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, Ilbc) { + auto factory = CreateAudioEncoderFactory<AudioEncoderIlbc>(); + EXPECT_THAT(factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ(AudioCodecInfo(8000, 1, 13333), + factory->QueryAudioEncoder({"ilbc", 8000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 8000, 1}, absl::nullopt)); + auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, absl::nullopt); + ASSERT_NE(nullptr, enc); + EXPECT_EQ(8000, enc->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, L16) { + auto factory = CreateAudioEncoderFactory<AudioEncoderL16>(); + EXPECT_THAT( + factory->GetSupportedEncoders(), + ::testing::ElementsAre( + AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}}, + AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}}, + AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}}, + AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}}, + AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}}, + AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0})); + EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16), + factory->QueryAudioEncoder({"L16", 48000, 1})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"L16", 8000, 0}, absl::nullopt)); + auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, absl::nullopt); + ASSERT_NE(nullptr, enc); + EXPECT_EQ(48000, enc->SampleRateHz()); +} + +TEST(AudioEncoderFactoryTemplateTest, Opus) { + auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>(); + AudioCodecInfo info = {48000, 1, 32000, 6000, 510000}; + info.allow_comfort_noise = false; + info.supports_network_adaption = true; + EXPECT_THAT( + factory->GetSupportedEncoders(), + ::testing::ElementsAre(AudioCodecSpec{ + {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}, + info})); + EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1})); + EXPECT_EQ( + info, + factory->QueryAudioEncoder( + {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}})); + EXPECT_EQ(nullptr, + factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt)); + auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, absl::nullopt); + ASSERT_NE(nullptr, enc); + EXPECT_EQ(48000, enc->SampleRateHz()); +} + +} // namespace webrtc |