Index: webrtc/modules/audio_coding/BUILD.gn |
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn |
index a60baba5da3efc05520704df9175fb09a2ae2746..e374abb674c25b5a1d68bd34adf6346cb78056bd 100644 |
--- a/webrtc/modules/audio_coding/BUILD.gn |
+++ b/webrtc/modules/audio_coding/BUILD.gn |
@@ -709,6 +709,8 @@ source_set("audio_network_adaptor") { |
"audio_network_adaptor/controller.h", |
"audio_network_adaptor/controller_manager.cc", |
"audio_network_adaptor/controller_manager.h", |
+ "audio_network_adaptor/debug_dump_writer.cc", |
+ "audio_network_adaptor/debug_dump_writer.h", |
"audio_network_adaptor/dtx_controller.cc", |
"audio_network_adaptor/dtx_controller.h", |
"audio_network_adaptor/frame_length_controller.cc", |
@@ -719,6 +721,13 @@ source_set("audio_network_adaptor") { |
] |
configs += [ "../..:common_config" ] |
public_configs = [ "../..:common_inherited_config" ] |
+ |
+ if (rtc_enable_protobuf) { |
+ deps = [ |
+ ":ana_debug_dump_proto", |
+ ] |
+ defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ] |
+ } |
} |
config("neteq_config") { |
@@ -975,6 +984,13 @@ if (rtc_include_tests) { |
} # audio_decoder_unittests |
if (rtc_enable_protobuf) { |
+ proto_library("ana_debug_dump_proto") { |
+ sources = [ |
+ "audio_network_adaptor/debug_dump.proto", |
+ ] |
+ proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor" |
+ } |
+ |
proto_library("neteq_unittest_proto") { |
sources = [ |
"neteq/neteq_unittest.proto", |