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-rw-r--r--third_party/libwebrtc/api/audio_codecs/BUILD.gn144
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diff --git a/third_party/libwebrtc/api/audio_codecs/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/BUILD.gn
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+++ b/third_party/libwebrtc/api/audio_codecs/BUILD.gn
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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",
+ "..: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",
+ ]
+}