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", |