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Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."

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This is a reland of https://codereview.webrtc.org/1737593002/ minus
the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and
webrtc/common.gyp that breaks GN in Chromium since it's using
the --check flag (which we should support).

BUG=webrtc:4243, webrtc:5589
TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully.
TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1740873003 .

Cr-Commit-Position: refs/heads/master@{#11794}
This commit is contained in:
kjellander@webrtc.org 2016-02-26 22:46:09 +01:00
parent f0fcbf3d83
commit 7ffeab525c
17 changed files with 20 additions and 14 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/api/remoteaudiosource.h
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@ -16,7 +16,7 @@
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/notifier.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/audio_sink.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"

1
webrtc/audio/webrtc_audio.gypi
View File

@ -18,7 +18,6 @@
'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',

6
webrtc/audio/audio_sink.h → webrtc/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_AUDIO_SINK_H_
#define WEBRTC_AUDIO_AUDIO_SINK_H_
#ifndef WEBRTC_AUDIO_SINK_H_
#define WEBRTC_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_AUDIO_SINK_H_
#endif // WEBRTC_AUDIO_SINK_H_

1
webrtc/common.gyp
View File

@ -12,6 +12,7 @@
'target_name': 'webrtc_common',
'type': 'static_library',
'sources': [
'audio_sink.h',
'common_types.cc',
'common_types.h',
'config.h',

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

@ -18,7 +18,7 @@
#include <string>
#include <vector>
#include "webrtc/audio/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/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

@ -60,9 +60,12 @@ 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

@ -12,6 +12,7 @@
'<(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

@ -99,6 +99,7 @@ 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/audio_sink.h"
#include "webrtc/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/audio_sink.h"
#include "webrtc/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/voice_engine/channel.h"

1
webrtc/voice_engine/voice_engine.gyp
View File

@ -15,6 +15,7 @@
'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',