| # 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_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", |
| "../..:webrtc_common", |
| "../../rtc_base:checks", |
| "../../rtc_base:deprecation", |
| "../../rtc_base:rtc_base_approved", |
| "../../rtc_base:sanitizer", |
| "../../rtc_base/system:rtc_export", |
| "//third_party/abseil-cpp/absl/strings", |
| "//third_party/abseil-cpp/absl/types:optional", |
| ] |
| } |
| |
| rtc_static_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", |
| "../../rtc_base:rtc_base_approved", |
| "L16:audio_decoder_L16", |
| "g711:audio_decoder_g711", |
| "g722:audio_decoder_g722", |
| "isac:audio_decoder_isac", |
| ] |
| 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_opus" ] |
| defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] |
| } else { |
| defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] |
| } |
| } |
| |
| rtc_static_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", |
| "../../rtc_base:rtc_base_approved", |
| "L16:audio_encoder_L16", |
| "g711:audio_encoder_g711", |
| "g722:audio_encoder_g722", |
| "isac:audio_encoder_isac", |
| ] |
| 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_opus" ] |
| defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ] |
| } else { |
| defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ] |
| } |
| } |