Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(21)

Unified Diff: webrtc/pc/BUILD.gn

Issue 2854123003: Build WebRTC with data channel only. (Closed)
Patch Set: Rebase. Created 3 years, 6 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/media/engine/webrtcvoiceengine.h ('k') | webrtc/pc/channel.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/pc/BUILD.gn
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index ae95a2cad300d74e9b56e5ed5082000346233d74..2720a1e8a86cf3b9dd3108109e432bf0beb9ca57 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -25,7 +25,7 @@ config("rtc_pc_config") {
}
}
-rtc_static_library("rtc_pc") {
+rtc_static_library("rtc_pc_base") {
defines = []
sources = [
"audiomonitor.cc",
@@ -59,8 +59,9 @@ rtc_static_library("rtc_pc") {
"../api:libjingle_peerconnection_api",
"../api:ortc_api",
"../base:rtc_base",
- "../common_video:common_video",
- "../media",
+ "../base:rtc_task_queue",
+ "../media:rtc_data",
+ "../media:rtc_media_base",
"../p2p:rtc_p2p",
]
@@ -76,6 +77,16 @@ rtc_static_library("rtc_pc") {
}
}
+rtc_source_set("rtc_pc") {
+ public_deps = [
+ ":rtc_pc_base",
+ ]
+
+ deps = [
+ "../media:rtc_audio_video",
+ ]
+}
+
config("libjingle_peerconnection_warnings_config") {
# GN orders flags on a target before flags from configs. The default config
# adds these flags so to cancel them out they need to come from a config and
@@ -85,7 +96,7 @@ config("libjingle_peerconnection_warnings_config") {
}
}
-rtc_static_library("libjingle_peerconnection") {
+rtc_static_library("peerconnection") {
cflags = []
sources = [
"audiotrack.cc",
@@ -146,19 +157,17 @@ rtc_static_library("libjingle_peerconnection") {
}
deps = [
- ":rtc_pc",
+ ":rtc_pc_base",
"..:webrtc_common",
"../api:call_api",
"../api:rtc_stats_api",
- "../api/audio_codecs:builtin_audio_decoder_factory",
- "../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../base:rtc_base",
"../base:rtc_base_approved",
- "../call",
+ "../call:call_interfaces",
"../logging:rtc_event_log_api",
- "../media",
- "../modules/audio_device:audio_device",
+ "../media:rtc_data",
+ "../media:rtc_media_base",
"../p2p:rtc_p2p",
"../stats",
"../system_wrappers:system_wrappers",
@@ -167,6 +176,48 @@ rtc_static_library("libjingle_peerconnection") {
public_deps = [
"../api:libjingle_peerconnection_api",
]
+}
+
+# This target implements CreatePeerConnectionFactory methods that will create a
+# PeerConnection will full functionality (audio, video and data). Applications
+# that wish to reduce their binary size by ommitting functionality they don't
+# need should use CreateModularCreatePeerConnectionFactory instead, using the
+# "peerconnection" build target and other targets specific to their
+# requrements. See comment in peerconnectionfactoryinterface.h.
+rtc_source_set("create_pc_factory") {
+ sources = [
+ "createpeerconnectionfactory.cc",
+ ]
+
+ deps = [
+ "../api:audio_mixer_api",
+ "../api:libjingle_peerconnection_api",
+ "../api/audio_codecs:audio_codecs_api",
+ "../api/audio_codecs:builtin_audio_decoder_factory",
+ "../api/audio_codecs:builtin_audio_encoder_factory",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
+ "../call",
+ "../call:call_interfaces",
+ "../logging:rtc_event_log_api",
+ "../media:rtc_audio_video",
+ "../modules/audio_device:audio_device",
+ ]
+
+ configs += [ ":libjingle_peerconnection_warnings_config" ]
+
+ 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" ]
+ }
+}
+
+rtc_source_set("libjingle_peerconnection") {
+ public_deps = [
+ ":create_pc_factory",
+ ":peerconnection",
+ "../api:libjingle_peerconnection_api",
+ ]
if (rtc_use_quic) {
sources += [
@@ -271,6 +322,8 @@ if (rtc_include_tests) {
"../base:rtc_base",
"../base:rtc_base_approved",
"../base:rtc_base_tests_utils",
+ "../call:call_interfaces",
+ "../logging:rtc_event_log_api",
"../media:rtc_media",
"../media:rtc_media_tests_utils",
"../modules/audio_device:audio_device",
« no previous file with comments | « webrtc/media/engine/webrtcvoiceengine.h ('k') | webrtc/pc/channel.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698