| Index: webrtc/modules/audio_coding/BUILD.gn
|
| diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
|
| index 7bbcd3aee2d40eaccdfbdafb6cae6daba9731e8a..d9a1a0247bfe2e0d8f91fb21847146c0e5617083 100644
|
| --- a/webrtc/modules/audio_coding/BUILD.gn
|
| +++ b/webrtc/modules/audio_coding/BUILD.gn
|
| @@ -9,6 +9,39 @@
|
| import("//build/config/arm.gni")
|
| import("../../build/webrtc.gni")
|
|
|
| +source_set("rent_a_codec") {
|
| + sources = [
|
| + "main/acm2/acm_codec_database.cc",
|
| + "main/acm2/acm_codec_database.h",
|
| + "main/acm2/rent_a_codec.cc",
|
| + "main/acm2/rent_a_codec.h",
|
| + ]
|
| + configs += [ "../..:common_config" ]
|
| + public_configs = [ "../..:common_inherited_config" ]
|
| + deps = [
|
| + "../..:webrtc_common",
|
| + ]
|
| +
|
| + defines = []
|
| + if (rtc_include_opus) {
|
| + defines += [ "WEBRTC_CODEC_OPUS" ]
|
| + }
|
| + if (!build_with_mozilla) {
|
| + if (current_cpu == "arm") {
|
| + defines += [ "WEBRTC_CODEC_ISACFX" ]
|
| + } else {
|
| + defines += [ "WEBRTC_CODEC_ISAC" ]
|
| + }
|
| + defines += [ "WEBRTC_CODEC_G722" ]
|
| + }
|
| + if (!build_with_mozilla && !build_with_chromium) {
|
| + defines += [
|
| + "WEBRTC_CODEC_ILBC",
|
| + "WEBRTC_CODEC_RED",
|
| + ]
|
| + }
|
| +}
|
| +
|
| config("audio_coding_config") {
|
| include_dirs = [
|
| "main/interface",
|
| @@ -18,8 +51,6 @@ config("audio_coding_config") {
|
|
|
| source_set("audio_coding") {
|
| sources = [
|
| - "main/acm2/acm_codec_database.cc",
|
| - "main/acm2/acm_codec_database.h",
|
| "main/acm2/acm_common_defs.h",
|
| "main/acm2/acm_receiver.cc",
|
| "main/acm2/acm_receiver.h",
|
| @@ -69,6 +100,7 @@ source_set("audio_coding") {
|
| ":g711",
|
| ":neteq",
|
| ":pcm16b",
|
| + ":rent_a_codec",
|
| "../..:rtc_event_log",
|
| "../..:webrtc_common",
|
| "../../common_audio",
|
|
|