| Index: webrtc/logging/BUILD.gn
|
| diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
|
| index ffc64a368fe50e0ac15628ac70511e3e5c821933..8bb6c265e4a3ded50fbb146df0a2c58d360c4412 100644
|
| --- a/webrtc/logging/BUILD.gn
|
| +++ b/webrtc/logging/BUILD.gn
|
| @@ -28,8 +28,8 @@ rtc_source_set("rtc_event_log_api") {
|
| ]
|
| deps = [
|
| "..:video_stream_api",
|
| + "..:webrtc_common",
|
| "../base:rtc_base_approved",
|
| - "../call:call_interfaces",
|
| ]
|
| }
|
|
|
| @@ -48,7 +48,6 @@ rtc_static_library("rtc_event_log_impl") {
|
| "..:webrtc_common",
|
| "../base:protobuf_utils",
|
| "../base:rtc_base_approved",
|
| - "../call:call_interfaces",
|
| "../modules/audio_coding:audio_network_adaptor",
|
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
|
| "../modules/rtp_rtcp",
|
| @@ -83,7 +82,6 @@ if (rtc_enable_protobuf) {
|
| ":rtc_event_log_api",
|
| ":rtc_event_log_proto",
|
| "..:webrtc_common",
|
| - "../call:call_interfaces",
|
| "../modules/audio_coding:audio_network_adaptor",
|
| "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
|
| "../modules/rtp_rtcp:rtp_rtcp",
|
| @@ -138,7 +136,6 @@ if (rtc_enable_protobuf) {
|
| ":rtc_event_log_impl",
|
| ":rtc_event_log_parser",
|
| "../base:rtc_base_approved",
|
| - "../call:call_interfaces",
|
| "../modules/rtp_rtcp:rtp_rtcp",
|
| "../system_wrappers:field_trial_default",
|
| "../system_wrappers:metrics_default",
|
| @@ -162,7 +159,6 @@ if (rtc_enable_protobuf) {
|
| ":rtc_event_log_impl",
|
| ":rtc_event_log_parser",
|
| "../base:rtc_base_approved",
|
| - "../call:call_interfaces",
|
|
|
| # TODO(kwiberg): Remove this dependency.
|
| "../api/audio_codecs:audio_codecs_api",
|
|
|