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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/audio_codecs/g711
parentInitial commit. (diff)
downloadthunderbird-upstream.tar.xz
thunderbird-upstream.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/api/audio_codecs/g711')
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn55
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc67
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h49
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build225
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc95
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h54
-rw-r--r--third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build225
7 files changed, 770 insertions, 0 deletions
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")