diff --git a/webrtc/DEPS b/webrtc/DEPS index 5fab15ccfd..41e0b712ce 100644 --- a/webrtc/DEPS +++ b/webrtc/DEPS @@ -9,11 +9,7 @@ include_rules = [ "+libyuv", "+testing", "-webrtc", # Has to be disabled; otherwise all dirs below will be allowed. - # Individual headers that will be moved out of here, see webrtc: - "+webrtc/audio_receive_stream.h", - "+webrtc/audio_send_stream.h", - "+webrtc/audio_sink.h", - "+webrtc/audio_state.h", + # Individual headers that will be moved out of here, see webrtc:4243. "+webrtc/call.h", "+webrtc/common.h", "+webrtc/common_types.h", @@ -29,20 +25,15 @@ include_rules = [ "+webrtc/video_send_stream.h", "+WebRTC", + "+webrtc/api", "+webrtc/base", "+webrtc/modules/include", "+webrtc/test", "+webrtc/tools", ] -# The below rules will be removed when webrtc: is fixed. +# The below rules will be removed when webrtc:4243 is fixed. specific_include_rules = { - "audio_send_stream\.h": [ - "+webrtc/modules/audio_coding", - ], - "audio_receive_stream\.h": [ - "+webrtc/modules/audio_coding/codecs/audio_decoder_factory.h", - ], "video_frame\.h": [ "+webrtc/common_video", ], diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index f09c4e4655..d14fe1f19e 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -19,6 +19,25 @@ group("api") { ] } +source_set("call_api") { + sources = [ + "call/audio_receive_stream.h", + "call/audio_send_stream.h", + "call/audio_sink.h", + "call/audio_state.h", + ] + + configs += [ "..:common_config" ] + public_configs = [ "..:common_inherited_config" ] + + deps = [ + # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. + "..:webrtc_common", + "../base:rtc_base_approved", + "../modules/audio_coding:audio_encoder_interface", + ] +} + 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 @@ -113,6 +132,7 @@ source_set("libjingle_peerconnection") { } deps = [ + ":call_api", "../call", "../media", "../pc", diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp index 274b87c792..5b14fdfc2d 100644 --- a/webrtc/api/api.gyp +++ b/webrtc/api/api.gyp @@ -94,10 +94,27 @@ }], ], # conditions 'targets': [ + { + 'target_name': 'call_api', + 'type': 'static_library', + 'dependencies': [ + # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done. + '<(webrtc_root)/base/base.gyp:rtc_base_approved', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface', + ], + 'sources': [ + 'call/audio_receive_stream.h', + 'call/audio_send_stream.h', + 'call/audio_sink.h', + 'call/audio_state.h', + ], + }, { 'target_name': 'libjingle_peerconnection', 'type': 'static_library', 'dependencies': [ + ':call_api', '<(webrtc_root)/media/media.gyp:rtc_media', '<(webrtc_root)/pc/pc.gyp:rtc_pc', ], diff --git a/webrtc/api/call/DEPS b/webrtc/api/call/DEPS new file mode 100644 index 0000000000..d1d4309681 --- /dev/null +++ b/webrtc/api/call/DEPS @@ -0,0 +1,4 @@ +include_rules = [ + "+webrtc/modules/audio_coding/codecs", +] + diff --git a/webrtc/audio_receive_stream.h b/webrtc/api/call/audio_receive_stream.h similarity index 96% rename from webrtc/audio_receive_stream.h rename to webrtc/api/call/audio_receive_stream.h index e0e9536a58..096cbc775e 100644 --- a/webrtc/audio_receive_stream.h +++ b/webrtc/api/call/audio_receive_stream.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_ -#define WEBRTC_AUDIO_RECEIVE_STREAM_H_ +#ifndef WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ +#define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ #include #include @@ -136,4 +136,4 @@ class AudioReceiveStream { }; } // namespace webrtc -#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_ +#endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_ diff --git a/webrtc/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h similarity index 96% rename from webrtc/audio_send_stream.h rename to webrtc/api/call/audio_send_stream.h index c3d0d339de..b309f7a221 100644 --- a/webrtc/audio_send_stream.h +++ b/webrtc/api/call/audio_send_stream.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_SEND_STREAM_H_ -#define WEBRTC_AUDIO_SEND_STREAM_H_ +#ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ +#define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ #include #include @@ -116,4 +116,4 @@ class AudioSendStream { }; } // namespace webrtc -#endif // WEBRTC_AUDIO_SEND_STREAM_H_ +#endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_ diff --git a/webrtc/audio_sink.h b/webrtc/api/call/audio_sink.h similarity index 92% rename from webrtc/audio_sink.h rename to webrtc/api/call/audio_sink.h index 2c932c5ab8..e865ead365 100644 --- a/webrtc/audio_sink.h +++ b/webrtc/api/call/audio_sink.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_SINK_H_ -#define WEBRTC_AUDIO_SINK_H_ +#ifndef WEBRTC_API_CALL_AUDIO_SINK_H_ +#define WEBRTC_API_CALL_AUDIO_SINK_H_ #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS) // Avoid conflict with format_macros.h. @@ -50,4 +50,4 @@ class AudioSinkInterface { } // namespace webrtc -#endif // WEBRTC_AUDIO_SINK_H_ +#endif // WEBRTC_API_CALL_AUDIO_SINK_H_ diff --git a/webrtc/audio_state.h b/webrtc/api/call/audio_state.h similarity index 92% rename from webrtc/audio_state.h rename to webrtc/api/call/audio_state.h index fa5784c844..ac912773aa 100644 --- a/webrtc/audio_state.h +++ b/webrtc/api/call/audio_state.h @@ -7,8 +7,8 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_AUDIO_STATE_H_ -#define WEBRTC_AUDIO_STATE_H_ +#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_ +#define WEBRTC_API_CALL_AUDIO_STATE_H_ #include "webrtc/base/refcount.h" #include "webrtc/base/scoped_ref_ptr.h" @@ -45,4 +45,4 @@ class AudioState : public rtc::RefCountInterface { }; } // namespace webrtc -#endif // WEBRTC_AUDIO_STATE_H_ +#endif // WEBRTC_API_CALL_AUDIO_STATE_H_ diff --git a/webrtc/api/remoteaudiosource.h b/webrtc/api/remoteaudiosource.h index 4cc68f8047..a67b89553e 100644 --- a/webrtc/api/remoteaudiosource.h +++ b/webrtc/api/remoteaudiosource.h @@ -14,8 +14,8 @@ #include #include +#include "webrtc/api/call/audio_sink.h" #include "webrtc/api/notifier.h" -#include "webrtc/audio_sink.h" #include "webrtc/base/criticalsection.h" #include "webrtc/pc/channel.h" diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index f8a8f67836..df4a48a7aa 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -17,12 +17,12 @@ #include #include +#include "webrtc/api/call/audio_sink.h" #include "webrtc/api/jsepicecandidate.h" #include "webrtc/api/jsepsessiondescription.h" #include "webrtc/api/peerconnectioninterface.h" #include "webrtc/api/sctputils.h" #include "webrtc/api/webrtcsessiondescriptionfactory.h" -#include "webrtc/audio_sink.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/bind.h" #include "webrtc/base/checks.h" diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index d031f69e10..762eb5f86e 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -31,6 +31,7 @@ source_set("audio") { deps = [ "..:webrtc_common", + "../api:call_api", "../system_wrappers", "../voice_engine", ] diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index ca2c08b34f..91237c341b 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -13,7 +13,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/audio/audio_state.h" #include "webrtc/audio/conversion.h" #include "webrtc/base/checks.h" diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index 24924c9a6d..871d48d6b0 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -13,8 +13,8 @@ #include -#include "webrtc/audio_receive_stream.h" -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_receive_stream.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_checker.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h index a993d5f2f9..ec2a4db413 100644 --- a/webrtc/audio/audio_send_stream.h +++ b/webrtc/audio/audio_send_stream.h @@ -13,8 +13,8 @@ #include -#include "webrtc/audio_send_stream.h" -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_send_stream.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_checker.h" #include "webrtc/call/bitrate_allocator.h" diff --git a/webrtc/audio/audio_state.h b/webrtc/audio/audio_state.h index 6bee3c6b99..31892d04fa 100644 --- a/webrtc/audio/audio_state.h +++ b/webrtc/audio/audio_state.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_AUDIO_AUDIO_STATE_H_ #define WEBRTC_AUDIO_AUDIO_STATE_H_ -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/audio/scoped_voe_interface.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/criticalsection.h" diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi index 9b4879a70b..6230057e4f 100644 --- a/webrtc/audio/webrtc_audio.gypi +++ b/webrtc/audio/webrtc_audio.gypi @@ -8,6 +8,7 @@ { 'variables': { 'webrtc_audio_dependencies': [ + '<(webrtc_root)/api/api.gyp:call_api', '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', diff --git a/webrtc/call.h b/webrtc/call.h index ec43b18ad4..5f64590506 100644 --- a/webrtc/call.h +++ b/webrtc/call.h @@ -13,13 +13,13 @@ #include #include -#include "webrtc/common_types.h" -#include "webrtc/audio_receive_stream.h" -#include "webrtc/audio_send_stream.h" -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_receive_stream.h" +#include "webrtc/api/call/audio_send_stream.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/base/networkroute.h" #include "webrtc/base/platform_file.h" #include "webrtc/base/socket.h" +#include "webrtc/common_types.h" #include "webrtc/video_receive_stream.h" #include "webrtc/video_send_stream.h" diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn index 5b428f72e9..2f3ce8c715 100644 --- a/webrtc/call/BUILD.gn +++ b/webrtc/call/BUILD.gn @@ -28,6 +28,7 @@ source_set("call") { deps = [ "..:rtc_event_log", "..:webrtc_common", + "../api:call_api", "../audio", "../modules/congestion_controller", "../modules/rtp_rtcp", diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc index 10d435d7d4..ab8643d62e 100644 --- a/webrtc/call/bitrate_estimator_tests.cc +++ b/webrtc/call/bitrate_estimator_tests.cc @@ -14,7 +14,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/base/checks.h" #include "webrtc/base/event.h" #include "webrtc/base/logging.h" diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc index fb6cac11bc..4c2562d75e 100644 --- a/webrtc/call/call_unittest.cc +++ b/webrtc/call/call_unittest.cc @@ -13,7 +13,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/call.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" #include "webrtc/test/mock_voice_engine.h" diff --git a/webrtc/common.gyp b/webrtc/common.gyp index c9ac71c898..e5d41da7c4 100644 --- a/webrtc/common.gyp +++ b/webrtc/common.gyp @@ -12,7 +12,6 @@ 'target_name': 'webrtc_common', 'type': 'static_library', 'sources': [ - 'audio_sink.h', 'common.cc', 'common.h', 'common_types.cc', diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn index 78e7f78f4d..df4e3d2d00 100644 --- a/webrtc/media/BUILD.gn +++ b/webrtc/media/BUILD.gn @@ -204,6 +204,7 @@ source_set("rtc_media") { deps += [ "..:webrtc_common", + "../api:call_api", "../base:rtc_base_approved", "../call", "../libjingle/xmllite", diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h index dfc7a1c66a..45603178ed 100644 --- a/webrtc/media/base/fakemediaengine.h +++ b/webrtc/media/base/fakemediaengine.h @@ -18,7 +18,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/copyonwritebuffer.h" #include "webrtc/base/networkroute.h" #include "webrtc/base/stringutils.h" diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h index 559617f536..866898cbfa 100644 --- a/webrtc/media/base/mediaengine.h +++ b/webrtc/media/base/mediaengine.h @@ -18,7 +18,7 @@ #include #include -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/api/rtpparameters.h" #include "webrtc/base/fileutils.h" #include "webrtc/base/sigslotrepeater.h" diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc index fdf7cf36fc..d0655cf3b0 100644 --- a/webrtc/media/engine/fakewebrtccall.cc +++ b/webrtc/media/engine/fakewebrtccall.cc @@ -13,7 +13,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/checks.h" #include "webrtc/base/gunit.h" #include "webrtc/media/base/rtputils.h" diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h index 8581d829d6..f723d7e8b5 100644 --- a/webrtc/media/engine/fakewebrtccall.h +++ b/webrtc/media/engine/fakewebrtccall.h @@ -23,8 +23,8 @@ #include #include -#include "webrtc/audio_receive_stream.h" -#include "webrtc/audio_send_stream.h" +#include "webrtc/api/call/audio_receive_stream.h" +#include "webrtc/api/call/audio_send_stream.h" #include "webrtc/base/buffer.h" #include "webrtc/call.h" #include "webrtc/video_frame.h" diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc index d0ca9aea61..0169d5a23a 100644 --- a/webrtc/media/engine/webrtcvoiceengine.cc +++ b/webrtc/media/engine/webrtcvoiceengine.cc @@ -18,7 +18,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/base64.h" #include "webrtc/base/byteorder.h" diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h index b2f9f622fc..6508e23896 100644 --- a/webrtc/media/engine/webrtcvoiceengine.h +++ b/webrtc/media/engine/webrtcvoiceengine.h @@ -16,7 +16,7 @@ #include #include -#include "webrtc/audio_state.h" +#include "webrtc/api/call/audio_state.h" #include "webrtc/base/buffer.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/networkroute.h" diff --git a/webrtc/media/media.gyp b/webrtc/media/media.gyp index c9482262d9..2eb72710d9 100644 --- a/webrtc/media/media.gyp +++ b/webrtc/media/media.gyp @@ -203,6 +203,7 @@ 'dependencies': [ '<(DEPTH)/testing/gmock.gyp:gmock', '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/api/api.gyp:call_api', '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils', '<(webrtc_root)/media/media.gyp:rtc_media', ], diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn index ea3e5ae807..906d99575b 100644 --- a/webrtc/pc/BUILD.gn +++ b/webrtc/pc/BUILD.gn @@ -49,6 +49,7 @@ source_set("rtc_pc") { ] deps = [ + "../api:call_api", "../base:rtc_base", "../media", ] diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc index 98838dd01c..f22bdc4ca4 100644 --- a/webrtc/pc/channel.cc +++ b/webrtc/pc/channel.cc @@ -12,7 +12,7 @@ #include "webrtc/pc/channel.h" -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/bind.h" #include "webrtc/base/byteorder.h" #include "webrtc/base/checks.h" diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h index 7d3bc946f6..6ea54194fb 100644 --- a/webrtc/pc/channel.h +++ b/webrtc/pc/channel.h @@ -18,7 +18,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/asyncinvoker.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/criticalsection.h" diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc index 60051a775e..2a23c94a7d 100644 --- a/webrtc/tools/event_log_visualizer/analyzer.cc +++ b/webrtc/tools/event_log_visualizer/analyzer.cc @@ -17,8 +17,8 @@ #include #include -#include "webrtc/audio_receive_stream.h" -#include "webrtc/audio_send_stream.h" +#include "webrtc/api/call/audio_receive_stream.h" +#include "webrtc/api/call/audio_send_stream.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/call.h" diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index 8f0e9f322f..ffa17ed266 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -92,6 +92,7 @@ source_set("voice_engine") { ":level_indicator", "..:rtc_event_log", "..:webrtc_common", + "../api:call_api", "../base:rtc_base_approved", "../common_audio", "../modules/audio_coding", diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index 6fc4f34040..adeb1dac2e 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -13,7 +13,7 @@ #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/optional.h" #include "webrtc/common_audio/resampler/include/push_resampler.h" diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc index 8c942f3701..fc22180566 100644 --- a/webrtc/voice_engine/channel_proxy.cc +++ b/webrtc/voice_engine/channel_proxy.cc @@ -12,7 +12,7 @@ #include -#include "webrtc/audio_sink.h" +#include "webrtc/api/call/audio_sink.h" #include "webrtc/base/checks.h" #include "webrtc/voice_engine/channel.h" diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 912b5228c1..dd6de61a52 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -15,6 +15,7 @@ 'target_name': 'voice_engine', 'type': 'static_library', 'dependencies': [ + '<(webrtc_root)/api/api.gyp:call_api', '<(webrtc_root)/base/base.gyp:rtc_base_approved', '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 8a56fe765c..a4c1c1d45f 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -17,9 +17,6 @@ 'target_name': 'webrtc', 'type': 'static_library', 'sources': [ - 'audio_receive_stream.h', - 'audio_send_stream.h', - 'audio_state.h', 'call.h', 'config.h', 'transport.h', diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi index 6d8ff6a145..93c6c88ca7 100644 --- a/webrtc/webrtc_tests.gypi +++ b/webrtc/webrtc_tests.gypi @@ -385,6 +385,7 @@ 'dependencies': [ '<(DEPTH)/testing/gmock.gyp:gmock', '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/api/api.gyp:call_api', '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', '<(webrtc_root)/modules/modules.gyp:video_capture',