From 7324eb9e62d1bd0a38c290ab36459a08e1f9b829 Mon Sep 17 00:00:00 2001 From: kjellander Date: Thu, 25 Feb 2016 08:36:42 -0800 Subject: [PATCH] Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) Reason for revert: Breaks GN in chromium. Original issue's description: > Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. > > webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is > depending on voice engine, resulting in a cyclic dependency (which we > don't detect since we have that check turned off, see webrtc:4243). > > BUG=webrtc:4243, webrtc:5589 > R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org > TBR=tommi@webrtc.org > > Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2 > Cr-Commit-Position: refs/heads/master@{#11766} TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4243, webrtc:5589 Review URL: https://codereview.webrtc.org/1739783002 Cr-Commit-Position: refs/heads/master@{#11769} --- webrtc/BUILD.gn | 16 +++------------- webrtc/api/remoteaudiosource.h | 2 +- webrtc/api/webrtcsession.cc | 2 +- webrtc/audio/audio_receive_stream.cc | 2 +- webrtc/{ => audio}/audio_sink.h | 6 +++--- webrtc/audio/webrtc_audio.gypi | 1 + webrtc/base/BUILD.gn | 1 - webrtc/base/base.gyp | 1 - webrtc/common.gyp | 10 +--------- webrtc/media/base/fakemediaengine.h | 2 +- webrtc/media/engine/fakewebrtccall.cc | 2 +- webrtc/media/engine/webrtcvoiceengine.cc | 2 +- webrtc/pc/channel.cc | 2 +- webrtc/pc/channel.h | 2 +- webrtc/video/BUILD.gn | 3 --- webrtc/video/webrtc_video.gypi | 1 - webrtc/voice_engine/BUILD.gn | 1 - webrtc/voice_engine/channel.h | 2 +- webrtc/voice_engine/channel_proxy.cc | 2 +- webrtc/voice_engine/voice_engine.gyp | 1 - webrtc/webrtc.gyp | 12 +++++++++--- 21 files changed, 27 insertions(+), 46 deletions(-) rename webrtc/{ => audio}/audio_sink.h (93%) diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index 1517d85c6b..6d84c504a3 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -167,12 +167,10 @@ config("common_config") { source_set("webrtc") { sources = [ - "audio_send_stream.h", - "audio_state.h", "call.h", - "video_decoder.h", - "video_encoder.h", - "video_frame.h", + "config.h", + "frame_callback.h", + "transport.h", ] defines = [] @@ -230,20 +228,12 @@ if (!build_with_chromium) { source_set("webrtc_common") { sources = [ - "audio_receive_stream.h", - "audio_sink.h", "common_types.cc", "common_types.h", "config.cc", "config.h", "engine_configurations.h", - "frame_callback.h", - "stream.h", - "transport.h", "typedefs.h", - "video_receive_stream.h", - "video_renderer.h", - "video_send_stream.h", ] configs += [ ":common_config" ] diff --git a/webrtc/api/remoteaudiosource.h b/webrtc/api/remoteaudiosource.h index 72ed17c58f..20e5d90cdd 100644 --- a/webrtc/api/remoteaudiosource.h +++ b/webrtc/api/remoteaudiosource.h @@ -16,7 +16,7 @@ #include "webrtc/api/mediastreaminterface.h" #include "webrtc/api/notifier.h" -#include "webrtc/audio_sink.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/criticalsection.h" #include "webrtc/media/base/audiorenderer.h" diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index e5cea14439..b2494140b4 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -23,7 +23,7 @@ #include "webrtc/api/peerconnectioninterface.h" #include "webrtc/api/sctputils.h" #include "webrtc/api/webrtcsessiondescriptionfactory.h" -#include "webrtc/audio_sink.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/checks.h" #include "webrtc/base/helpers.h" diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 9c25389471..2c58def560 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/audio/audio_sink.h" #include "webrtc/audio/audio_state.h" #include "webrtc/audio/conversion.h" #include "webrtc/base/checks.h" diff --git a/webrtc/audio_sink.h b/webrtc/audio/audio_sink.h similarity index 93% rename from webrtc/audio_sink.h rename to webrtc/audio/audio_sink.h index 2c932c5ab8..999644f4ce 100644 --- a/webrtc/audio_sink.h +++ b/webrtc/audio/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_AUDIO_AUDIO_SINK_H_ +#define WEBRTC_AUDIO_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_AUDIO_AUDIO_SINK_H_ diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi index 9b4879a70b..53b7d16b1a 100644 --- a/webrtc/audio/webrtc_audio.gypi +++ b/webrtc/audio/webrtc_audio.gypi @@ -18,6 +18,7 @@ 'audio/audio_receive_stream.h', 'audio/audio_send_stream.cc', 'audio/audio_send_stream.h', + 'audio/audio_sink.h', 'audio/audio_state.cc', 'audio/audio_state.h', 'audio/conversion.h', diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn index 54c78b399f..a056269c95 100644 --- a/webrtc/base/BUILD.gn +++ b/webrtc/base/BUILD.gn @@ -126,7 +126,6 @@ static_library("rtc_base_approved") { "event_tracer.h", "exp_filter.cc", "exp_filter.h", - "format_macros.h", "md5.cc", "md5.h", "md5digest.cc", diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp index 29912544d5..3c569fae47 100644 --- a/webrtc/base/base.gyp +++ b/webrtc/base/base.gyp @@ -94,7 +94,6 @@ 'event_tracer.h', 'exp_filter.cc', 'exp_filter.h', - 'format_macros.h', 'logging.cc', 'logging.h', 'md5.cc', diff --git a/webrtc/common.gyp b/webrtc/common.gyp index 9ca4bf4e22..3b5fe902dd 100644 --- a/webrtc/common.gyp +++ b/webrtc/common.gyp @@ -12,20 +12,12 @@ 'target_name': 'webrtc_common', 'type': 'static_library', 'sources': [ - 'audio_receive_stream.h', - 'audio_sink.h', 'common_types.cc', 'common_types.h', - 'config.cc', 'config.h', + 'config.cc', 'engine_configurations.h', - 'frame_callback.h', - 'stream.h', - 'transport.h', 'typedefs.h', - 'video_receive_stream.h', - 'video_renderer.h', - 'video_send_stream.h', ], }, ], diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h index d68db70830..f2de5ac653 100644 --- a/webrtc/media/base/fakemediaengine.h +++ b/webrtc/media/base/fakemediaengine.h @@ -17,7 +17,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/buffer.h" #include "webrtc/base/stringutils.h" #include "webrtc/media/base/audiorenderer.h" diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc index 93b565dbf3..af098af822 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/audio/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/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc index 1d235934da..3709e807d3 100644 --- a/webrtc/media/engine/webrtcvoiceengine.cc +++ b/webrtc/media/engine/webrtcvoiceengine.cc @@ -21,7 +21,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/arraysize.h" #include "webrtc/base/base64.h" #include "webrtc/base/byteorder.h" diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc index 447687a425..6a696ea4b4 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/audio/audio_sink.h" #include "webrtc/base/bind.h" #include "webrtc/base/buffer.h" #include "webrtc/base/byteorder.h" diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h index f72818924d..abecd669e5 100644 --- a/webrtc/pc/channel.h +++ b/webrtc/pc/channel.h @@ -17,7 +17,7 @@ #include #include -#include "webrtc/audio_sink.h" +#include "webrtc/audio/audio_sink.h" #include "webrtc/base/asyncudpsocket.h" #include "webrtc/base/criticalsection.h" #include "webrtc/base/network.h" diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 4f1b7ae197..e35772e22c 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -60,12 +60,9 @@ source_set("video") { deps = [ "..:rtc_event_log", "..:webrtc_common", - "../base:rtc_base_approved", "../common_video", "../modules/bitrate_controller", - "../modules/congestion_controller", "../modules/pacing", - "../modules/remote_bitrate_estimator", "../modules/rtp_rtcp", "../modules/utility", "../modules/video_capture:video_capture_module", diff --git a/webrtc/video/webrtc_video.gypi b/webrtc/video/webrtc_video.gypi index f11ce95727..db8d5c7e89 100644 --- a/webrtc/video/webrtc_video.gypi +++ b/webrtc/video/webrtc_video.gypi @@ -12,7 +12,6 @@ '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/common_video/common_video.gyp:common_video', '<(webrtc_root)/modules/modules.gyp:bitrate_controller', - '<(webrtc_root)/modules/modules.gyp:congestion_controller', '<(webrtc_root)/modules/modules.gyp:paced_sender', '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', '<(webrtc_root)/modules/modules.gyp:video_capture_module', diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index 13104c6c86..82cd92355c 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -99,7 +99,6 @@ source_set("voice_engine") { deps = [ "..:rtc_event_log", "..:webrtc_common", - "../base:rtc_base_approved", "../common_audio", "../modules/audio_coding", "../modules/audio_conference_mixer", diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index a3cd5d6535..0e87252877 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/audio/audio_sink.h" #include "webrtc/base/criticalsection.h" #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/common_types.h" diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc index da7864f15f..3beaf9b294 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/audio/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 cff2d8f2d9..ff588d8ead 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -15,7 +15,6 @@ 'target_name': 'voice_engine', 'type': 'static_library', 'dependencies': [ - '<(webrtc_root)/base/base.gyp:rtc_base_approved', '<(webrtc_root)/common.gyp:webrtc_common', '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', '<(webrtc_root)/modules/modules.gyp:audio_coding_module', diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index 0c87e42031..1adae73f9e 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -110,12 +110,18 @@ 'target_name': 'webrtc', 'type': 'static_library', 'sources': [ + 'audio_receive_stream.h', 'audio_send_stream.h', 'audio_state.h', 'call.h', - 'video_frame.h', - 'video_decoder.h', - 'video_encoder.h', + 'config.h', + 'frame_callback.h', + 'stream.h', + 'transport.h', + 'video_receive_stream.h', + 'video_renderer.h', + 'video_send_stream.h', + '<@(webrtc_audio_sources)', '<@(webrtc_call_sources)', '<@(webrtc_video_sources)',