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Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )

Browse Source 
Reason for revert:
The openmax_dl include change breaks downstream projects.

Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623

Review URL: https://codereview.webrtc.org/1808573002

Cr-Commit-Position: refs/heads/master@{#12009}
This commit is contained in:
kjellander
2016-03-15 17:41:01 -07:00
committed by Commit bot
parent 086f851b7b
commit 56cf60e717
43 changed files with 306 additions and 348 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
DEPS
View File

@ -23,6 +23,25 @@ deps_os = {
},
}
# Define rules for which include paths are allowed in our source.
include_rules = [
# Base is only used to build Android APK tests and may not be referenced by
# WebRTC production code.
'-base',
'-chromium',
'+external/webrtc/webrtc', # Android platform build.
'+gflags',
'+libyuv',
'+net',
'+talk',
'+testing',
'+third_party',
'+unicode',
'+usrsctplib',
'+webrtc',
'+vpx',
]
hooks = [
{
# Check for legacy named top-level dir (named 'trunk').

7
talk/app/webrtc/DEPS
View File

@ -1,7 +0,0 @@
include_rules = [
"+talk/app/webrtc/objc",
"+webrtc/video_frame.h",
"+webrtc/api",
"+webrtc/base",
"+webrtc/media",
]

47
webrtc/DEPS
View File

@ -1,47 +0,0 @@
# Define rules for which include paths are allowed in our source.
include_rules = [
# Base is only used to build Android APK tests and may not be referenced by
# WebRTC production code.
"-base",
"-chromium",
"+external/webrtc/webrtc", # Android platform build.
"+gflags",
"+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",
"+webrtc/call.h",
"+webrtc/common.h",
"+webrtc/common_types.h",
"+webrtc/config.h",
"+webrtc/engine_configurations.h",
"+webrtc/frame_callback.h",
"+webrtc/stream.h",
"+webrtc/transport.h",
"+webrtc/typedefs.h",
"+webrtc/video_decoder.h",
"+webrtc/video_encoder.h",
"+webrtc/video_frame.h",
"+webrtc/video_receive_stream.h",
"+webrtc/video_renderer.h",
"+webrtc/video_send_stream.h",
"+webrtc/base",
"+webrtc/modules/include",
"+webrtc/test",
"+webrtc/tools",
]
# The below rules will be removed when webrtc: is fixed.
specific_include_rules = {
"audio_send_stream\.h": [
"+webrtc/modules/audio_coding",
],
"video_frame\.h": [
"+webrtc/common_video",
],
}

23
webrtc/api/DEPS
View File

@ -1,23 +0,0 @@
include_rules = [
"+third_party/libyuv",
"+webrtc/base",
"+webrtc/common_video",
"+webrtc/media",
"+webrtc/p2p",
"+webrtc/pc",
"+webrtc/modules/audio_device",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/video_coding",
"+webrtc/modules/video_render",
"+webrtc/system_wrappers",
]
specific_include_rules = {
"androidtestinitializer\.cc": [
"+base/android", # Allowed only for Android tests.
"+webrtc/voice_engine",
],
"peerconnection_jni\.cc": [
"+webrtc/voice_engine",
]
}

10
webrtc/audio/DEPS
View File

@ -1,10 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/voice_engine",
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
"+webrtc/modules/pacing",
"+webrtc/modules/remote_bitrate_estimator",
"+webrtc/modules/rtp_rtcp",
"+webrtc/system_wrappers",
]

11
webrtc/base/DEPS
View File

@ -1,11 +0,0 @@
include_rules = [
"+json",
"+third_party/jsoncpp",
"+webrtc/system_wrappers",
]
specific_include_rules = {
"gunit_prod.h": [
"+gtest",
],
}

13
webrtc/call/DEPS
View File

@ -1,13 +0,0 @@
include_rules = [
"+webrtc/audio",
"+webrtc/base",
"+webrtc/modules/audio_coding",
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
"+webrtc/modules/pacing",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/utility",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
"+webrtc/video",
]

5
webrtc/common_audio/DEPS
View File

@ -1,5 +0,0 @@
include_rules = [
"+third_party/openmax_dl",
"+webrtc/base",
"+webrtc/system_wrappers",
]

2
webrtc/common_audio/real_fourier_openmax.cc
View File

@ -12,7 +12,7 @@
#include <cstdlib>
#include "third_party/openmax_dl/dl/sp/api/omxSP.h"
#include "dl/sp/api/omxSP.h"
#include "webrtc/base/checks.h"
namespace webrtc {

4
webrtc/common_video/DEPS
View File

@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
]

7
webrtc/examples/DEPS
View File

@ -1,7 +0,0 @@
include_rules = [
"+webrtc/api",
"+webrtc/base",
"+webrtc/media",
"+webrtc/modules/audio_device",
"+webrtc/p2p",
]

5
webrtc/libjingle/DEPS
View File

@ -1,5 +0,0 @@
include_rules = [
"+third_party/expat",
"+webrtc/base",
"+webrtc/p2p",
]

280
webrtc/libjingle/xmpp/chatroommodule_unittest.cc Normal file
View File

@ -0,0 +1,280 @@
/*
* Copyright 2004 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.
*/
#include <iostream>
#include <sstream>
#include <string>
#include "buzz/chatroommodule.h"
#include "buzz/constants.h"
#include "buzz/xmlelement.h"
#include "buzz/xmppengine.h"
#include "common/common.h"
#include "engine/util_unittest.h"
#include "test/unittest-inl.h"
#include "test/unittest.h"
#define TEST_OK(x) TEST_EQ((x),XMPP_RETURN_OK)
#define TEST_BADARGUMENT(x) TEST_EQ((x),XMPP_RETURN_BADARGUMENT)
namespace buzz {
class MultiUserChatModuleTest;
static void
WriteEnteredStatus(std::ostream& os, XmppChatroomEnteredStatus status) {
switch(status) {
case XMPP_CHATROOM_ENTERED_SUCCESS:
os<<"success";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_NICKNAME_CONFLICT:
os<<"failure(nickname conflict)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_REQUIRED:
os<<"failure(password required)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_PASSWORD_INCORRECT:
os<<"failure(password incorrect)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_NOT_A_MEMBER:
os<<"failure(not a member)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_MEMBER_BANNED:
os<<"failure(member banned)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_MAX_USERS:
os<<"failure(max users)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_ROOM_LOCKED:
os<<"failure(room locked)";
break;
case XMPP_CHATROOM_ENTERED_FAILURE_UNSPECIFIED:
os<<"failure(unspecified)";
break;
default:
os<<"unknown";
break;
}
}
static void
WriteExitedStatus(std::ostream& os, XmppChatroomExitedStatus status) {
switch (status) {
case XMPP_CHATROOM_EXITED_REQUESTED:
os<<"requested";
break;
case XMPP_CHATROOM_EXITED_BANNED:
os<<"banned";
break;
case XMPP_CHATROOM_EXITED_KICKED:
os<<"kicked";
break;
case XMPP_CHATROOM_EXITED_NOT_A_MEMBER:
os<<"not member";
break;
case XMPP_CHATROOM_EXITED_SYSTEM_SHUTDOWN:
os<<"system shutdown";
break;
case XMPP_CHATROOM_EXITED_UNSPECIFIED:
os<<"unspecified";
break;
default:
os<<"unknown";
break;
}
}
//! This session handler saves all calls to a string. These are events and
//! data delivered form the engine to application code.
class XmppTestChatroomHandler : public XmppChatroomHandler {
public:
XmppTestChatroomHandler() {}
virtual ~XmppTestChatroomHandler() {}
void ChatroomEnteredStatus(XmppChatroomModule* room,
XmppChatroomEnteredStatus status) {
RTC_UNUSED(room);
ss_ <<"[ChatroomEnteredStatus status: ";
WriteEnteredStatus(ss_, status);
ss_ <<"]";
}
void ChatroomExitedStatus(XmppChatroomModule* room,
XmppChatroomExitedStatus status) {
RTC_UNUSED(room);
ss_ <<"[ChatroomExitedStatus status: ";
WriteExitedStatus(ss_, status);
ss_ <<"]";
}
void MemberEntered(XmppChatroomModule* room,
const XmppChatroomMember* entered_member) {
RTC_UNUSED(room);
ss_ << "[MemberEntered " << entered_member->member_jid().Str() << "]";
}
void MemberExited(XmppChatroomModule* room,
const XmppChatroomMember* exited_member) {
RTC_UNUSED(room);
ss_ << "[MemberExited " << exited_member->member_jid().Str() << "]";
}
void MemberChanged(XmppChatroomModule* room,
const XmppChatroomMember* changed_member) {
RTC_UNUSED(room);
ss_ << "[MemberChanged " << changed_member->member_jid().Str() << "]";
}
virtual void MessageReceived(XmppChatroomModule* room, const XmlElement& message) {
RTC_UNUSED2(room, message);
}
std::string Str() {
return ss_.str();
}
std::string StrClear() {
std::string result = ss_.str();
ss_.str("");
return result;
}
private:
std::stringstream ss_;
};
//! This is the class that holds all of the unit test code for the
//! roster module
class XmppChatroomModuleTest : public UnitTest {
public:
XmppChatroomModuleTest() {}
void TestEnterExitChatroom() {
std::stringstream dump;
// Configure the engine
rtc::scoped_ptr<XmppEngine> engine(XmppEngine::Create());
XmppTestHandler handler(engine.get());
// Configure the module and handler
rtc::scoped_ptr<XmppChatroomModule> chatroom(XmppChatroomModule::Create());
// Configure the module handler
chatroom->RegisterEngine(engine.get());
// Set up callbacks
engine->SetOutputHandler(&handler);
engine->AddStanzaHandler(&handler);
engine->SetSessionHandler(&handler);
// Set up minimal login info
engine->SetUser(Jid("david@my-server"));
engine->SetPassword("david");
// Do the whole login handshake
RunLogin(this, engine.get(), &handler);
TEST_EQ("", handler.OutputActivity());
// Get the chatroom and set the handler
XmppTestChatroomHandler chatroom_handler;
chatroom->set_chatroom_handler(static_cast<XmppChatroomHandler*>(&chatroom_handler));
// try to enter the chatroom
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM);
chatroom->set_nickname("thirdwitch");
chatroom->set_chatroom_jid(Jid("darkcave@my-server"));
chatroom->RequestEnterChatroom("", XMPP_CONNECTION_STATUS_UNKNOWN, "en");
TEST_EQ(chatroom_handler.StrClear(), "");
TEST_EQ(handler.OutputActivity(),
"<presence to=\"darkcave@my-server/thirdwitch\">"
"<muc:x xmlns:muc=\"http://jabber.org/protocol/muc\"/>"
"</presence>");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
// simulate the server and test the client
std::string input;
input = "<presence from=\"darkcave@my-server/firstwitch\" to=\"david@my-server\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"owner\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(), "");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
input = "<presence from=\"darkcave@my-server/secondwitch\" to=\"david@my-server\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"member\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(), "");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_ENTER);
input = "<presence from=\"darkcave@my-server/thirdwitch\" to=\"david@my-server\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"member\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(),
"[ChatroomEnteredStatus status: success]");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
// simulate somebody else entering the room after we entered
input = "<presence from=\"darkcave@my-server/fourthwitch\" to=\"david@my-server\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"member\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(), "[MemberEntered darkcave@my-server/fourthwitch]");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
// simulate somebody else leaving the room after we entered
input = "<presence from=\"darkcave@my-server/secondwitch\" to=\"david@my-server\" type=\"unavailable\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"member\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(), "[MemberExited darkcave@my-server/secondwitch]");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_IN_ROOM);
// try to leave the room
chatroom->RequestExitChatroom();
TEST_EQ(chatroom_handler.StrClear(), "");
TEST_EQ(handler.OutputActivity(),
"<presence to=\"darkcave@my-server/thirdwitch\" type=\"unavailable\"/>");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_REQUESTED_EXIT);
// simulate the server and test the client
input = "<presence from=\"darkcave@my-server/thirdwitch\" to=\"david@my-server\" type=\"unavailable\">"
"<x xmlns=\"http://jabber.org/protocol/muc#user\">"
"<item affiliation=\"member\" role=\"participant\"/>"
"</x>"
"</presence>";
TEST_OK(engine->HandleInput(input.c_str(), input.length()));
TEST_EQ(chatroom_handler.StrClear(),
"[ChatroomExitedStatus status: requested]");
TEST_EQ(chatroom->state(), XMPP_CHATROOM_STATE_NOT_IN_ROOM);
}
};
// A global function that creates the test suite for this set of tests.
TestBase* ChatroomModuleTest_Create() {
TestSuite* suite = new TestSuite("ChatroomModuleTest");
ADD_TEST(suite, XmppChatroomModuleTest, TestEnterExitChatroom);
return suite;
}
}

22
webrtc/media/DEPS
View File

@ -1,22 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
"+webrtc/modules/audio_coding",
"+webrtc/modules/audio_device",
"+webrtc/modules/audio_processing",
"+webrtc/modules/video_capture",
"+webrtc/modules/video_coding",
"+webrtc/p2p",
"+webrtc/pc",
"+webrtc/sound",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
"+usrsctplib",
]
specific_include_rules = {
"win32devicemanager\.cc": [
"+third_party/logitech/files/logitechquickcam.h",
],
}

7
webrtc/modules/audio_coding/DEPS
View File

@ -1,7 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_audio",
"+webrtc/audio_coding/neteq/neteq_unittest.pb.h", # Different path.
"+webrtc/system_wrappers",
]

4
webrtc/modules/audio_conference_mixer/DEPS
View File

@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
]

11
webrtc/modules/audio_device/DEPS
View File

@ -1,11 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/system_wrappers",
]
specific_include_rules = {
"ensure_initialized\.cc": [
"+base/android",
],
}

14
webrtc/modules/audio_processing/DEPS
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@ -1,14 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/system_wrappers",
]
specific_include_rules = {
".*test\.cc": [
"+webrtc/tools",
# Android platform build has different paths.
"+gtest",
"+external/webrtc",
],
}

4
webrtc/modules/audio_processing/agc/agc_unittest.cc
View File

@ -10,8 +10,8 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"

2
webrtc/modules/audio_processing/agc/mock_agc.h
View File

@ -13,7 +13,7 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "gmock/gmock.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {

5
webrtc/modules/bitrate_controller/DEPS
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@ -1,5 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/system_wrappers",
]

5
webrtc/modules/congestion_controller/DEPS
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@ -1,5 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
"+webrtc/video",
]

4
webrtc/modules/desktop_capture/DEPS
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@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
]

4
webrtc/modules/include/DEPS
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@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_video",
]

5
webrtc/modules/media_file/DEPS
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@ -1,5 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/system_wrappers",
]

4
webrtc/modules/pacing/DEPS
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@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
]

10
webrtc/modules/remote_bitrate_estimator/DEPS
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@ -1,10 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/system_wrappers",
]
specific_include_rules = {
"nada\.h": [
"+webrtc/voice_engine",
],
}

6
webrtc/modules/rtp_rtcp/DEPS
View File

@ -1,6 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
"+webrtc/system_wrappers",
]

6
webrtc/modules/utility/DEPS
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@ -1,6 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/common_video",
"+webrtc/system_wrappers",
]

5
webrtc/modules/video_capture/DEPS
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@ -1,5 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_video",
"+webrtc/system_wrappers",
]

9
webrtc/modules/video_coding/DEPS
View File

@ -1,9 +0,0 @@
include_rules = [
"+third_party/ffmpeg",
"+third_party/openh264",
"+vpx",
"+webrtc/base",
"+webrtc/common_video",
"+webrtc/system_wrappers",
"+webrtc/tools",
]

2
webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
View File

@ -11,7 +11,7 @@
#include <memory>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "gtest/gtest.h"
#include "vpx/vpx_encoder.h"
#include "vpx/vp8cx.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"

3
webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h
View File

@ -15,7 +15,6 @@
#include <memory>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h"
@ -23,6 +22,8 @@
#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
#include "webrtc/video_frame.h"
#include "gtest/gtest.h"
using ::testing::_;
using ::testing::AllOf;
using ::testing::Field;

6
webrtc/modules/video_processing/DEPS
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@ -1,6 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_audio",
"+webrtc/common_video",
"+webrtc/system_wrappers",
]

5
webrtc/modules/video_render/DEPS
View File

@ -1,5 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_video",
"+webrtc/system_wrappers",
]

5
webrtc/p2p/DEPS
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@ -1,5 +0,0 @@
include_rules = [
"+net",
"+webrtc/base",
"+webrtc/system_wrappers",
]

13
webrtc/pc/DEPS
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@ -1,13 +0,0 @@
include_rules = [
"+webrtc/api",
"+webrtc/base",
"+webrtc/media",
"+webrtc/p2p",
"+third_party/libsrtp"
]
specific_include_rules = {
"srtpfilter_unittest\.cc": [
"+crypto",
],
}

4
webrtc/sound/DEPS
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@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
]

4
webrtc/system_wrappers/DEPS
View File

@ -1,4 +0,0 @@
include_rules = [
"+webrtc/base",
]

13
webrtc/test/DEPS
View File

@ -1,13 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
"+webrtc/modules/audio_coding",
"+webrtc/modules/audio_device",
"+webrtc/modules/media_file",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/video_capture",
"+webrtc/modules/video_coding",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
]

8
webrtc/tools/DEPS
View File

@ -1,8 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/common_video",
"+webrtc/modules/audio_processing",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
]

17
webrtc/video/DEPS
View File

@ -1,17 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_video",
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
"+webrtc/modules/pacing",
"+webrtc/modules/remote_bitrate_estimator",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/utility",
"+webrtc/modules/video_coding",
"+webrtc/modules/video_capture",
"+webrtc/modules/video_processing",
"+webrtc/modules/video_render",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
]

14
webrtc/voice_engine/DEPS
View File

@ -1,14 +0,0 @@
include_rules = [
"+webrtc/base",
"+webrtc/call",
"+webrtc/common_audio",
"+webrtc/modules/audio_coding",
"+webrtc/modules/audio_conference_mixer",
"+webrtc/modules/audio_device",
"+webrtc/modules/audio_processing",
"+webrtc/modules/media_file",
"+webrtc/modules/pacing",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/utility",
"+webrtc/system_wrappers",
]