| Index: webrtc/modules/audio_coding/BUILD.gn
|
| diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
|
| index 3a2d20a800e8326fe4b6077992474c7213ffd31d..6702b11ba6bb93d576a47d6b6e89ffe8b485b6c3 100644
|
| --- a/webrtc/modules/audio_coding/BUILD.gn
|
| +++ b/webrtc/modules/audio_coding/BUILD.gn
|
| @@ -82,6 +82,11 @@ rtc_static_library("rent_a_codec") {
|
| ":isac_fix_c",
|
| ":neteq_decoder_enum",
|
| ] + audio_codec_deps
|
| +
|
| + if (rtc_enable_protobuf) {
|
| + deps += [ "../../base:protobuf_utils" ]
|
| + }
|
| +
|
| defines = audio_codec_defines
|
| }
|
|
|
| @@ -837,6 +842,10 @@ rtc_static_library("webrtc_opus") {
|
| ":webrtc_opus_c",
|
| ]
|
|
|
| + if (rtc_enable_protobuf) {
|
| + deps += [ "../../base:protobuf_utils" ]
|
| + }
|
| +
|
| defines = audio_codec_defines
|
| if (rtc_opus_variable_complexity) {
|
| defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
|
| @@ -928,6 +937,7 @@ rtc_static_library("audio_network_adaptor") {
|
| deps += [
|
| ":ana_config_proto",
|
| ":ana_debug_dump_proto",
|
| + "../../base:protobuf_utils",
|
| ]
|
| defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
|
| }
|
| @@ -1190,6 +1200,11 @@ if (rtc_include_tests) {
|
| "../../system_wrappers:system_wrappers",
|
| "../../test:test_support",
|
| ]
|
| +
|
| + if (rtc_enable_protobuf) {
|
| + deps += [ "../../base:protobuf_utils" ]
|
| + }
|
| +
|
| if (!build_with_chromium && is_clang) {
|
| # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
|
| suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
| @@ -1329,6 +1344,10 @@ if (rtc_include_tests) {
|
| "//testing/gtest",
|
| ]
|
|
|
| + if (rtc_enable_protobuf) {
|
| + deps += [ "../../base:protobuf_utils" ]
|
| + }
|
| +
|
| data = audio_decoder_unittests_resources
|
|
|
| if (is_android) {
|
| @@ -2102,6 +2121,7 @@ if (rtc_include_tests) {
|
| deps += [
|
| ":ana_config_proto",
|
| ":neteq_unittest_proto",
|
| + "../../base:protobuf_utils",
|
| ]
|
| }
|
|
|
|
|