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Fr4nz D13trich 2025-11-22 14:04:28 +01:00
parent 81b91f4139
commit f8c34fa5ee
22732 changed files with 4815320 additions and 2 deletions
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147
TMessagesProj/jni/voip/webrtc/api/audio_codecs/BUILD.gn Normal file
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# 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",
"..:ref_count",
"..:scoped_refptr",
"../../api:field_trials_view",
"../../api:rtp_parameters",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:event_tracer",
"../../rtc_base:refcount",
"../../rtc_base:sanitizer",
"../../rtc_base/system:rtc_export",
"../units:data_rate",
"../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",
]
}