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Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )

Browse Source 
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}
This commit is contained in:
kjellander 2016-02-25 08:36:42 -08:00 committed by Commit bot
parent 3dd5d1d84a
commit 7324eb9e62
21 changed files with 27 additions and 46 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

16
webrtc/BUILD.gn
View File

@ -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" ]

2
webrtc/api/remoteaudiosource.h
View File

@ -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"

2
webrtc/api/webrtcsession.cc
View File

@ -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"

2
webrtc/audio/audio_receive_stream.cc
View File

@ -13,7 +13,7 @@
#include <string>
#include <utility>
#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"

6
webrtc/audio_sink.h → webrtc/audio/audio_sink.h
View File

@ -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_

1
webrtc/audio/webrtc_audio.gypi
View File

@ -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',

1
webrtc/base/BUILD.gn
View File

@ -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",

1
webrtc/base/base.gyp
View File

@ -94,7 +94,6 @@
'event_tracer.h',
'exp_filter.cc',
'exp_filter.h',
'format_macros.h',
'logging.cc',
'logging.h',
'md5.cc',

10
webrtc/common.gyp
View File

@ -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',
],
},
],

2
webrtc/media/base/fakemediaengine.h
View File

@ -17,7 +17,7 @@
#include <string>
#include <vector>
#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"

2
webrtc/media/engine/fakewebrtccall.cc
View File

@ -13,7 +13,7 @@
#include <algorithm>
#include <utility>
#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"

2
webrtc/media/engine/webrtcvoiceengine.cc
View File

@ -21,7 +21,7 @@
#include <string>
#include <vector>
#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"

2
webrtc/pc/channel.cc
View File

@ -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"

2
webrtc/pc/channel.h
View File

@ -17,7 +17,7 @@
#include <utility>
#include <vector>
#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"

3
webrtc/video/BUILD.gn
View File

@ -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",

1
webrtc/video/webrtc_video.gypi
View File

@ -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',

1
webrtc/voice_engine/BUILD.gn
View File

@ -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",

2
webrtc/voice_engine/channel.h
View File

@ -13,7 +13,7 @@
#include <memory>
#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"

2
webrtc/voice_engine/channel_proxy.cc
View File

@ -12,7 +12,7 @@
#include <utility>
#include "webrtc/audio_sink.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/voice_engine/channel.h"

1
webrtc/voice_engine/voice_engine.gyp
View File

@ -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',

12
webrtc/webrtc.gyp
View File

@ -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)',