From 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 Mon Sep 17 00:00:00 2001 From: Gustaf Ullberg Date: Fri, 16 Feb 2018 13:43:49 +0100 Subject: [PATCH] BUILD.gn file for api/audio. Targets containing files in api/audio are moved from api/BUILD.gn to api/audio/BUILD.gn. Bug: webrtc:8844 Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981 Reviewed-on: https://webrtc-review.googlesource.com/54360 Reviewed-by: Karl Wiberg Reviewed-by: Stefan Holmer Commit-Queue: Gustaf Ullberg Cr-Commit-Position: refs/heads/master@{#22074} --- api/BUILD.gn | 60 ++--------------------------- api/audio/BUILD.gn | 63 +++++++++++++++++++++++++++++++ audio/BUILD.gn | 2 +- call/BUILD.gn | 2 +- modules/BUILD.gn | 2 +- modules/audio_mixer/BUILD.gn | 4 +- modules/audio_processing/BUILD.gn | 6 +-- pc/BUILD.gn | 2 +- 8 files changed, 75 insertions(+), 66 deletions(-) create mode 100644 api/audio/BUILD.gn diff --git a/api/BUILD.gn b/api/BUILD.gn index 78a513729e..5eef0e11d0 100644 --- a/api/BUILD.gn +++ b/api/BUILD.gn @@ -29,10 +29,10 @@ rtc_source_set("call_api") { deps = [ # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. - ":audio_mixer_api", ":transport_api", "..:webrtc_common", "../rtc_base:rtc_base_approved", + "audio:audio_mixer_api", "audio_codecs:audio_codecs_api", ] } @@ -100,13 +100,13 @@ rtc_static_library("libjingle_peerconnection_api") { deps = [ ":array_view", - ":audio_mixer_api", ":audio_options_api", ":callfactory_api", ":libjingle_logging_api", ":optional", ":rtc_stats_api", ":video_frame_api", + "audio:audio_mixer_api", "audio_codecs:audio_codecs_api", # Basically, don't add stuff here. You might break sensitive downstream @@ -192,60 +192,6 @@ rtc_source_set("rtc_stats_api") { ] } -rtc_source_set("audio_frame_api") { - visibility = [ "*" ] - sources = [ - "audio/audio_frame.cc", - "audio/audio_frame.h", - ] - - deps = [ - "../:typedefs", - "../rtc_base:checks", - "../rtc_base:deprecation", - "../rtc_base:rtc_base_approved", - ] -} - -rtc_source_set("audio_mixer_api") { - visibility = [ "*" ] - sources = [ - "audio/audio_mixer.h", - ] - - deps = [ - ":audio_frame_api", - "../rtc_base:rtc_base_approved", - ] -} - -rtc_source_set("aec3_config") { - visibility = [ "*" ] - sources = [ - "audio/echo_canceller3_config.h", - ] -} - -rtc_source_set("aec3_factory") { - visibility = [ "*" ] - sources = [ - "audio/echo_canceller3_factory.cc", - "audio/echo_canceller3_factory.h", - ] - - deps = [ - ":aec3_config", - ":echo_control", - ] -} - -rtc_source_set("echo_control") { - visibility = [ "*" ] - sources = [ - "audio/echo_control.h", - ] -} - rtc_source_set("audio_options_api") { visibility = [ "*" ] sources = [ @@ -367,8 +313,8 @@ if (rtc_include_tests) { ] deps = [ - ":audio_mixer_api", "../test:test_support", + "audio:audio_mixer_api", ] } diff --git a/api/audio/BUILD.gn b/api/audio/BUILD.gn new file mode 100644 index 0000000000..9fb963a7df --- /dev/null +++ b/api/audio/BUILD.gn @@ -0,0 +1,63 @@ +# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. +# +# Use of this source code is governed by a BSD-style license +# that can be found in the LICENSE file in the root of the source +# tree. An additional intellectual property rights grant can be found +# in the file PATENTS. All contributing project authors may +# be found in the AUTHORS file in the root of the source tree. + +import("../../webrtc.gni") + +rtc_source_set("audio_frame_api") { + visibility = [ "*" ] + sources = [ + "audio_frame.cc", + "audio_frame.h", + ] + + deps = [ + "../../:typedefs", + "../../rtc_base:checks", + "../../rtc_base:deprecation", + "../../rtc_base:rtc_base_approved", + ] +} + +rtc_source_set("audio_mixer_api") { + visibility = [ "*" ] + sources = [ + "audio_mixer.h", + ] + + deps = [ + ":audio_frame_api", + "../../rtc_base:rtc_base_approved", + ] +} + +rtc_source_set("aec3_config") { + visibility = [ "*" ] + sources = [ + "echo_canceller3_config.h", + ] +} + +rtc_source_set("aec3_factory") { + visibility = [ "*" ] + sources = [ + "echo_canceller3_factory.cc", + "echo_canceller3_factory.h", + ] + + deps = [ + ":aec3_config", + ":echo_control", + ] +} + +rtc_source_set("echo_control") { + visibility = [ "*" ] + sources = [ + "echo_control.h", + ] +} diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 5b5c723c4f..cfd320a85b 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -47,11 +47,11 @@ rtc_static_library("audio") { deps = [ "..:webrtc_common", "../api:array_view", - "../api:audio_mixer_api", "../api:call_api", "../api:libjingle_peerconnection_api", "../api:optional", "../api:transport_api", + "../api/audio:audio_mixer_api", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_encoder_factory", "../call:bitrate_allocator", diff --git a/call/BUILD.gn b/call/BUILD.gn index 6be92fd7c9..993be206b8 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -26,11 +26,11 @@ rtc_source_set("call_interfaces") { ":video_stream_api", "..:webrtc_common", "../:typedefs", - "../api:audio_mixer_api", "../api:fec_controller_api", "../api:libjingle_peerconnection_api", "../api:optional", "../api:transport_api", + "../api/audio:audio_mixer_api", "../api/audio_codecs:audio_codecs_api", "../modules/audio_processing:audio_processing_statistics", "../rtc_base:rtc_base", diff --git a/modules/BUILD.gn b/modules/BUILD.gn index a45fd3487f..d912366183 100644 --- a/modules/BUILD.gn +++ b/modules/BUILD.gn @@ -51,11 +51,11 @@ rtc_source_set("module_api") { ":module_api_public", "..:webrtc_common", "../:typedefs", - "../api:audio_frame_api", "../api:libjingle_peerconnection_api", "../api:optional", "../api:video_frame_api", "../api:video_frame_api_i420", + "../api/audio:audio_frame_api", "../rtc_base:deprecation", "../rtc_base:rtc_base_approved", "video_coding:codec_globals_headers", diff --git a/modules/audio_mixer/BUILD.gn b/modules/audio_mixer/BUILD.gn index ce46e76169..2758dc72ba 100644 --- a/modules/audio_mixer/BUILD.gn +++ b/modules/audio_mixer/BUILD.gn @@ -38,7 +38,7 @@ rtc_static_library("audio_mixer_impl") { "../..:webrtc_common", "../../:typedefs", "../../api:array_view", - "../../api:audio_mixer_api", + "../../api/audio:audio_mixer_api", "../../audio/utility:audio_frame_operations", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", @@ -84,7 +84,7 @@ if (rtc_include_tests) { ":audio_mixer_impl", "..:module_api", "../../api:array_view", - "../../api:audio_mixer_api", + "../../api/audio:audio_mixer_api", "../../audio/utility:audio_frame_operations", "../../rtc_base:checks", "../../rtc_base:rtc_base_approved", diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn index a724ba40fa..617440262c 100644 --- a/modules/audio_processing/BUILD.gn +++ b/modules/audio_processing/BUILD.gn @@ -221,10 +221,10 @@ rtc_static_library("audio_processing") { "..:module_api", "../..:webrtc_common", "../../:typedefs", - "../../api:aec3_config", "../../api:array_view", - "../../api:echo_control", "../../api:optional", + "../../api/audio:aec3_config", + "../../api/audio:echo_control", "../../audio/utility:audio_frame_operations", "../../common_audio:common_audio_c", "../../rtc_base:checks", @@ -585,9 +585,9 @@ if (rtc_include_tests) { "..:module_api", "../..:webrtc_common", "../../:typedefs", - "../../api:aec3_config", "../../api:array_view", "../../api:optional", + "../../api/audio:aec3_config", "../../common_audio:common_audio", "../../common_audio:common_audio_c", "../../rtc_base:checks", diff --git a/pc/BUILD.gn b/pc/BUILD.gn index 21be8575be..0d0c511c58 100644 --- a/pc/BUILD.gn +++ b/pc/BUILD.gn @@ -214,9 +214,9 @@ rtc_static_library("create_pc_factory") { ] deps = [ - "../api:audio_mixer_api", "../api:callfactory_api", "../api:libjingle_peerconnection_api", + "../api/audio:audio_mixer_api", "../api/audio_codecs:audio_codecs_api", "../api/video_codecs:video_codecs_api", "../call",