Build WebRTC with data channel only.

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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")
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }
 
 group("pc") {
   public_deps = [
     ":rtc_pc",
   ]
 }
 
 config("rtc_pc_config") {
   defines = []
   if (rtc_enable_sctp) {
     defines += [ "HAVE_SCTP" ]
   }
 }
 
-rtc_static_library("rtc_pc") {
+rtc_static_library("rtc_pc_audio") {
+  sources = [
+    "channel_audio.cc",
+  ]
+
+  deps = [
+    "../media:rtc_audio_video",
+  ]
+}
+
+rtc_static_library("rtc_pc_audio_nullimpl") {
+  sources = [
+    "channel_audio_nullimpl.cc",
+  ]
+}
+
+rtc_static_library("rtc_pc_base") {
   defines = []
   sources = [
     "audiomonitor.cc",
     "audiomonitor.h",
     "bundlefilter.cc",
     "bundlefilter.h",
     "channel.cc",
     "channel.h",
     "channelmanager.cc",
     "channelmanager.h",
@@ skipping 10 matching lines @@
     "rtptransport.cc",
     "rtptransport.h",
     "srtpfilter.cc",
     "srtpfilter.h",
     "voicechannel.h",
   ]
 
   deps = [
     "../api:call_api",
     "../base:rtc_base",
-    "../media",
+    "../media:rtc_media",
   ]
 
   if (rtc_build_libsrtp) {
     deps += [ "//third_party/libsrtp" ]
   }
 
   public_configs = [ ":rtc_pc_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" ]
   }
 }
 
+# TODO(zhihuang): Remove this once the downstream dependencies start using the
+# modular targets.
+rtc_static_library("rtc_pc") {
+  deps = [
+    ":rtc_pc_audio",
+    ":rtc_pc_base",
+  ]
+}
+
 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
   # cannot be on the target directly.
   if (!is_win && !is_clang) {
     cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
   }
 }
 
+rtc_static_library("libjingle_peerconnection_audio") {

[Zhi Huang, 2017/05/18 03:55:38]

Use this when building with audio.

[pthatcher, 2017/05/18 17:50:26]

Perhaps a better name would be "audio".

[Zhi Huang, 2017/05/23 03:40:35]

We already have a target named "audio" and I'll name it "webrtc_audio". Similar
for others.

+  sources = [
+    "peerconnectionfactory_audio.cc",
+  ]
+
+  public_deps = [
+    ":rtc_pc_audio",
+  ]
+}
+
+rtc_static_library("libjingle_peerconnection_audio_nullimpl") {
+  sources = [
+    "peerconnectionfactory_audio_nullimpl.cc",
+  ]
+
+  public_deps = [
+    ":rtc_pc_audio_nullimpl",
+  ]
+}
+

[pthatcher, 2017/05/18 17:50:26]

And "null_audio"

+rtc_static_library("libjingle_peerconnection_media_nullimpl") {
+  sources = [
+    "peerconnection_media_nullimpl.cc",
+    "peerconnectionfactory_media_nullimpl.cc",
+  ]
+}
+
+rtc_static_library("libjingle_peerconnection_media") {

[Zhi Huang, 2017/05/18 03:55:38]

Use this when build with either audio or video.

+  sources = [
+    "peerconnection_media.cc",
+    "peerconnectionfactory_media.cc",
+  ]
+
+  deps = [
+    "../call",
+  ]
+}
+
+rtc_static_library("libjingle_peerconnection_base") {

[Zhi Huang, 2017/05/18 03:55:38]

The target without audio and video.

[pthatcher, 2017/05/18 17:50:26]

This could be just "base"

+  cflags = []
+  sources = [
+    "audiotrack.cc",
+    "audiotrack.h",
+    "datachannel.cc",
+    "datachannel.h",
+    "dtmfsender.cc",
+    "dtmfsender.h",
+    "iceserverparsing.cc",
+    "iceserverparsing.h",
+    "jsepicecandidate.cc",
+    "jsepsessiondescription.cc",
+    "localaudiosource.cc",
+    "localaudiosource.h",
+    "mediastream.cc",
+    "mediastream.h",
+    "mediastreamobserver.cc",
+    "mediastreamobserver.h",
+    "mediastreamtrack.h",
+    "peerconnection.cc",
+    "peerconnection.h",
+    "peerconnectionfactory.cc",
+    "peerconnectionfactory.h",
+    "remoteaudiosource.cc",
+    "remoteaudiosource.h",
+    "rtcstatscollector.cc",
+    "rtcstatscollector.h",
+    "rtpreceiver.cc",
+    "rtpreceiver.h",
+    "rtpsender.cc",
+    "rtpsender.h",
+    "sctputils.cc",
+    "sctputils.h",
+    "statscollector.cc",
+    "statscollector.h",
+    "streamcollection.h",
+    "trackmediainfomap.cc",
+    "trackmediainfomap.h",
+    "videocapturertracksource.cc",
+    "videocapturertracksource.h",
+    "videotrack.cc",
+    "videotrack.h",
+    "videotracksource.cc",
+    "videotracksource.h",
+    "webrtcsdp.cc",
+    "webrtcsdp.h",
+    "webrtcsession.cc",
+    "webrtcsession.h",
+    "webrtcsessiondescriptionfactory.cc",
+    "webrtcsessiondescriptionfactory.h",
+  ]
+
+  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" ]
+  }
+
+  deps = [
+    ":rtc_pc_base",
+    "../api:call_api",
+    "../api:rtc_stats_api",
+    "../api/video_codecs:video_codecs_api",
+    "../logging:rtc_event_log_api",
+    "../stats",
+  ]
+
+  public_deps = [
+    "../api:libjingle_peerconnection_api",
+  ]
+}
+
+# TODO(zhihuang): Remove this once the downstream dependencies start using the
+# modular targets.

[Taylor Brandstetter, 2017/05/18 17:57:05]

Couldn't this target just be defined in terms of the modular targets? Having a
single build target that includes everything is a nice convenience.

[Zhi Huang, 2017/05/23 03:40:35]

Done.

 rtc_static_library("libjingle_peerconnection") {
   cflags = []
   sources = [
     "audiotrack.cc",
     "audiotrack.h",
     "datachannel.cc",
     "datachannel.h",
     "dtmfsender.cc",
     "dtmfsender.h",
     "iceserverparsing.cc",
@@ skipping 269 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }