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Delete unneeded includes of rtc_base/stringutils.h.

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Also delete corresponding dependencies on rtc_base:stringutils.

Bug: webrtc:6424
Change-Id: I2be5e021292eea2d788c76a63cc0e4f7cefd927d
Reviewed-on: https://webrtc-review.googlesource.com/c/114544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26057}
This commit is contained in:
Niels Möller 2018-12-19 11:26:37 +01:00 committed by Commit Bot
parent aabac232b1
commit 31d8b52075
27 changed files with 0 additions and 33 deletions
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1
api/BUILD.gn
View File

@ -158,7 +158,6 @@ rtc_static_library("libjingle_peerconnection_api") {
"../rtc_base:deprecation",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_base/system:rtc_export",
]

1
api/video_codecs/video_codec.cc
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@ -15,7 +15,6 @@
#include "absl/strings/match.h"
#include "rtc_base/checks.h"
#include "rtc_base/stringutils.h"
namespace webrtc {

1
audio/BUILD.gn
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@ -78,7 +78,6 @@ rtc_static_library("audio") {
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_minmax",
"../rtc_base:stringutils",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",

3
examples/BUILD.gn
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@ -698,7 +698,6 @@ if (is_linux || is_win) {
"../media:rtc_media_base",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:stringutils",
"../rtc_base/third_party/sigslot",
"../system_wrappers:field_trial",
"../test:field_trial",
@ -763,7 +762,6 @@ if (is_linux || is_win) {
]
deps = [
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_tools:command_line_parser",
"../system_wrappers:field_trial",
"../test:field_trial",
@ -933,7 +931,6 @@ if (!build_with_chromium) {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
]
}
}

1
media/engine/webrtcvideoengine.cc
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@ -32,7 +32,6 @@
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/timeutils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"

1
media/engine/webrtcvideoengine_unittest.cc
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@ -49,7 +49,6 @@
#include "rtc_base/fakeclock.h"
#include "rtc_base/gunit.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/stringutils.h"
#include "test/field_trial.h"
#include "test/gmock.h"

1
media/sctp/sctptransport.cc
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@ -36,7 +36,6 @@ enum PreservedErrno {
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/thread_checker.h"
#include "rtc_base/trace_event.h"
#include "usrsctplib/usrsctp.h"

2
modules/audio_processing/BUILD.gn
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@ -331,7 +331,6 @@ rtc_source_set("apm_logging") {
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:stringutils",
]
defines = []
}
@ -583,7 +582,6 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_json",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:stringutils",
"../../system_wrappers",
"../../test:test_support",
"aec_dump",

1
modules/audio_processing/test/audio_processing_simulator.cc
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@ -31,7 +31,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/strings/json.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/stringutils.h"
namespace webrtc {
namespace test {

1
modules/rtp_rtcp/BUILD.gn
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@ -215,7 +215,6 @@ rtc_static_library("rtp_rtcp") {
"../../rtc_base:rtc_numerics",
"../../rtc_base:safe_minmax",
"../../rtc_base:sequenced_task_checker",
"../../rtc_base:stringutils",
"../../rtc_base/system:fallthrough",
"../../rtc_base/time:timestamp_extrapolator",
"../../system_wrappers",

1
modules/rtp_rtcp/source/rtp_utility.cc
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@ -24,7 +24,6 @@
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringutils.h"
namespace webrtc {

2
p2p/BUILD.gn
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@ -92,7 +92,6 @@ rtc_static_library("rtc_p2p") {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:safe_minmax",
"../rtc_base:stringutils",
"../rtc_base:weak_ptr",
"../rtc_base/network:sent_packet",
"../rtc_base/system:rtc_export",
@ -184,7 +183,6 @@ if (rtc_include_tests) {
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:stringutils",
"../rtc_base:testclient",
"../rtc_base/network:sent_packet",
"../rtc_base/third_party/sigslot:sigslot",

1
p2p/base/dtlstransport_unittest.cc
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@ -24,7 +24,6 @@
#include "rtc_base/ssladapter.h"
#include "rtc_base/sslidentity.h"
#include "rtc_base/sslstreamadapter.h"
#include "rtc_base/stringutils.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(rtc::SSLStreamAdapter::feature())) { \

3
pc/BUILD.gn
View File

@ -90,7 +90,6 @@ rtc_static_library("rtc_pc_base") {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_task_queue",
"../rtc_base:stringutils",
"../rtc_base/third_party/base64",
"../rtc_base/third_party/sigslot",
"../system_wrappers:metrics",
@ -214,7 +213,6 @@ rtc_static_library("peerconnection") {
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:stringutils",
"../rtc_base/experiments:congestion_controller_experiment",
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/base64",
@ -520,7 +518,6 @@ if (rtc_include_tests) {
"../rtc_base:checks",
"../rtc_base:gunit_helpers",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:stringutils",
"../rtc_base/third_party/base64",
"../rtc_base/third_party/sigslot:sigslot",
"../system_wrappers:metrics",

1
pc/mediasession.cc
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@ -31,7 +31,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/third_party/base64/base64.h"
namespace {

1
pc/peerconnection_signaling_unittest.cc
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@ -29,7 +29,6 @@
#include "pc/test/fakeaudiocapturemodule.h"
#include "pc/test/fakertccertificategenerator.h"
#include "rtc_base/gunit.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/virtualsocketserver.h"
#include "test/gmock.h"

1
pc/peerconnectionendtoend_unittest.cc
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@ -21,7 +21,6 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringutils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"

1
pc/rtcstatscollector_unittest.cc
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@ -35,7 +35,6 @@
#include "rtc_base/fakesslidentity.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/timeutils.h"
using testing::AtLeast;

1
pc/webrtcsdp.cc
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@ -45,7 +45,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/messagedigest.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/stringutils.h"
using cricket::AudioContentDescription;
using cricket::Candidate;

1
rtc_base/httpcommon.cc
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@ -31,7 +31,6 @@
#include "rtc_base/messagedigest.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/third_party/base64/base64.h" // for Base64
#include "rtc_base/zero_memory.h" // for ExplicitZeroMemory

1
rtc_base/macutils.cc
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@ -16,7 +16,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/macutils.h"
#include "rtc_base/stringutils.h"
namespace rtc {

1
rtc_base/opensslstreamadapter.cc
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@ -34,7 +34,6 @@
#include "rtc_base/opensslidentity.h"
#include "rtc_base/sslcertificate.h"
#include "rtc_base/stream.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/thread.h"
#include "rtc_base/timeutils.h"

1
rtc_base/socketadapters.cc
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@ -37,7 +37,6 @@
#include "rtc_base/logging.h"
#include "rtc_base/socketadapters.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/zero_memory.h"
namespace rtc {

1
sdk/android/BUILD.gn
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@ -762,7 +762,6 @@ if (is_android) {
"../../rtc_base:rtc_base",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:stringutils",
"../../system_wrappers:field_trial",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",

1
sdk/android/src/jni/pc/peerconnectionfactory.cc
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@ -22,7 +22,6 @@
// The user may pass in a nullptr.
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
#include "rtc_base/event_tracer.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/thread.h"
#include "sdk/android/generated_peerconnection_jni/jni/PeerConnectionFactory_jni.h"
#include "sdk/android/native_api/jni/java_types.h"

1
test/scenario/BUILD.gn
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@ -87,7 +87,6 @@ if (rtc_include_tests) {
"../../rtc_base:rtc_task_queue",
"../../rtc_base:safe_minmax",
"../../rtc_base:sequenced_task_checker",
"../../rtc_base:stringutils",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../video",

1
test/scenario/scenario_tests/BUILD.gn
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@ -23,7 +23,6 @@ if (rtc_include_tests) {
"../../:test_support",
"../../../modules/congestion_controller/bbr",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:stringutils",
"../../../rtc_base/experiments:field_trial_parser",
"//testing/gtest",
]