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Fixing WebRTC after moving from src/webrtc to src/

Browse source 
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
This commit is contained in:
Mirko Bonadei 2017-09-15 06:47:31 +02:00 committed by Commit Bot
parent bb547203bf
commit 92ea95e34a
3635 changed files with 19692 additions and 19645 deletions
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8
.gitignore vendored
View file

@ -57,9 +57,9 @@
/tools_webrtc/video_quality_toolchain/mac/zxing
/tools_webrtc/video_quality_toolchain/win/*.dll
/tools_webrtc/video_quality_toolchain/win/*.exe
/webrtc/rtc_tools/testing/*.zip
/webrtc/rtc_tools/testing/*.gz
/webrtc/rtc_tools/testing/golang/*/*.gz
/webrtc/rtc_tools/testing/golang/*/*.zip
/rtc_tools/testing/*.zip
/rtc_tools/testing/*.gz
/rtc_tools/testing/golang/*/*.gz
/rtc_tools/testing/golang/*/*.zip
/x86-generic_out/
/xcodebuild

24
.gn
View file

@ -20,7 +20,29 @@ secondary_source = "//build/secondary/"
# matching these patterns (see "gn help label_pattern" for format) will have
# their includes checked for proper dependencies when you run either
# "gn check" or "gn gen --check".
check_targets = [ "//webrtc/*" ]
check_targets = [
"//api/*",
"//audio/*",
"//backup/*",
"//call/*",
"//common_audio/*",
"//common_video/*",
"//examples/*",
"//logging/*",
"//media/*",
"//modules/*",
"//ortc/*",
"//p2p/*",
"//pc/*",
"//rtc_base/*",
"//rtc_tools/*",
"//sdk/*",
"//stats/*",
"//system_wrappers/*",
"//test/*",
"//video/*",
"//voice_engine/*",
]
# These are the list of GN files that run exec_script. This whitelist exists
# to force additional review for new uses of exec_script, which is strongly

30
BUILD.gn
View file

@ -431,8 +431,8 @@ if (rtc_include_tests) {
# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
video_engine_tests_resources = [
"../resources/foreman_cif_short.yuv",
"../resources/voice_engine/audio_long16.pcm",
"resources/foreman_cif_short.yuv",
"resources/voice_engine/audio_long16.pcm",
]
if (is_ios) {
@ -475,19 +475,19 @@ if (rtc_include_tests) {
}
webrtc_perf_tests_resources = [
"../resources/audio_coding/speech_mono_16kHz.pcm",
"../resources/audio_coding/speech_mono_32_48kHz.pcm",
"../resources/audio_coding/testfile32kHz.pcm",
"../resources/ConferenceMotion_1280_720_50.yuv",
"../resources/difficult_photo_1850_1110.yuv",
"../resources/foreman_cif.yuv",
"../resources/google-wifi-3mbps.rx",
"../resources/paris_qcif.yuv",
"../resources/photo_1850_1110.yuv",
"../resources/presentation_1850_1110.yuv",
"../resources/verizon4g-downlink.rx",
"../resources/voice_engine/audio_long16.pcm",
"../resources/web_screenshot_1850_1110.yuv",
"resources/audio_coding/speech_mono_16kHz.pcm",
"resources/audio_coding/speech_mono_32_48kHz.pcm",
"resources/audio_coding/testfile32kHz.pcm",
"resources/ConferenceMotion_1280_720_50.yuv",
"resources/difficult_photo_1850_1110.yuv",
"resources/foreman_cif.yuv",
"resources/google-wifi-3mbps.rx",
"resources/paris_qcif.yuv",
"resources/photo_1850_1110.yuv",
"resources/presentation_1850_1110.yuv",
"resources/verizon4g-downlink.rx",
"resources/voice_engine/audio_long16.pcm",
"resources/web_screenshot_1850_1110.yuv",
]
if (is_ios) {

26
DEPS
View file

@ -113,7 +113,7 @@ deps_os = {
'src/third_party/ub-uiautomator/lib':
Var('chromium_git') + '/chromium/third_party/ub-uiautomator.git' + '@' + '00270549ce3161ae72ceb24712618ea28b4f9434',
# Gradle 3.5.0. Used for testing Android Studio project generation for WebRTC.
'src/webrtc/examples/androidtests/third_party/gradle':
'src/examples/androidtests/third_party/gradle':
Var('chromium_git') + '/external/github.com/gradle/gradle.git' + '@' +
'941559e020f6c357ebb08d5c67acdb858a3defc2',
},
@ -554,26 +554,26 @@ include_rules = [
"+libyuv",
"-webrtc", # Has to be disabled; otherwise all dirs below will be allowed.
# Individual headers that will be moved out of here, see webrtc:4243.
"+webrtc/call/rtp_config.h",
"+webrtc/common_types.h",
"+webrtc/transport.h",
"+webrtc/typedefs.h",
"+webrtc/voice_engine_configurations.h",
"+call/rtp_config.h",
"+common_types.h",
"+transport.h",
"+typedefs.h",
"+voice_engine_configurations.h",
"+WebRTC",
"+webrtc/api",
"+webrtc/modules/include",
"+webrtc/rtc_base",
"+webrtc/test",
"+webrtc/rtc_tools",
"+api",
"+modules/include",
"+rtc_base",
"+test",
"+rtc_tools",
]
# The below rules will be removed when webrtc:4243 is fixed.
specific_include_rules = {
"video_receive_stream\.h": [
"+webrtc/call/video_receive_stream.h",
"+call/video_receive_stream.h",
],
"video_send_stream\.h": [
"+webrtc/call/video_send_stream.h",
"+call/video_send_stream.h",
],
}

139
PRESUBMIT.py
View file

@ -15,29 +15,29 @@ import sys
# Files and directories that are *skipped* by cpplint in the presubmit script.
CPPLINT_BLACKLIST = [
'api/video_codecs/video_decoder.h',
'common_types.cc',
'common_types.h',
'examples/objc',
'media',
'modules/audio_coding',
'modules/audio_conference_mixer',
'modules/audio_device',
'modules/audio_processing',
'modules/desktop_capture',
'modules/include/module_common_types.h',
'modules/media_file',
'modules/utility',
'modules/video_capture',
'p2p',
'pc',
'rtc_base',
'sdk/android/src/jni',
'sdk/objc',
'system_wrappers',
'test',
'tools_webrtc',
'webrtc/api/video_codecs/video_decoder.h',
'webrtc/examples/objc',
'webrtc/media',
'webrtc/modules/audio_coding',
'webrtc/modules/audio_conference_mixer',
'webrtc/modules/audio_device',
'webrtc/modules/audio_processing',
'webrtc/modules/desktop_capture',
'webrtc/modules/include/module_common_types.h',
'webrtc/modules/media_file',
'webrtc/modules/utility',
'webrtc/modules/video_capture',
'webrtc/p2p',
'webrtc/pc',
'webrtc/rtc_base',
'webrtc/sdk/android/src/jni',
'webrtc/sdk/objc',
'webrtc/system_wrappers',
'webrtc/test',
'webrtc/voice_engine',
'webrtc/common_types.h',
'webrtc/common_types.cc',
'voice_engine',
]
# These filters will always be removed, even if the caller specifies a filter
@ -62,34 +62,33 @@ BLACKLIST_LINT_FILTERS = [
# webrtc-users@google.com (internal list).
# 4. (later) The deprecated APIs are removed.
NATIVE_API_DIRS = (
'webrtc',
'webrtc/api',
'webrtc/media',
'webrtc/modules/audio_device/include',
'webrtc/pc',
'api',
'media',
'modules/audio_device/include',
'pc',
)
# These directories should not be used but are maintained only to avoid breaking
# some legacy downstream code.
LEGACY_API_DIRS = (
'webrtc/common_audio/include',
'webrtc/modules/audio_coding/include',
'webrtc/modules/audio_conference_mixer/include',
'webrtc/modules/audio_processing/include',
'webrtc/modules/bitrate_controller/include',
'webrtc/modules/congestion_controller/include',
'webrtc/modules/include',
'webrtc/modules/remote_bitrate_estimator/include',
'webrtc/modules/rtp_rtcp/include',
'webrtc/modules/rtp_rtcp/source',
'webrtc/modules/utility/include',
'webrtc/modules/video_coding/codecs/h264/include',
'webrtc/modules/video_coding/codecs/i420/include',
'webrtc/modules/video_coding/codecs/vp8/include',
'webrtc/modules/video_coding/codecs/vp9/include',
'webrtc/modules/video_coding/include',
'webrtc/rtc_base',
'webrtc/system_wrappers/include',
'webrtc/voice_engine/include',
'common_audio/include',
'modules/audio_coding/include',
'modules/audio_conference_mixer/include',
'modules/audio_processing/include',
'modules/bitrate_controller/include',
'modules/congestion_controller/include',
'modules/include',
'modules/remote_bitrate_estimator/include',
'modules/rtp_rtcp/include',
'modules/rtp_rtcp/source',
'modules/utility/include',
'modules/video_coding/codecs/h264/include',
'modules/video_coding/codecs/i420/include',
'modules/video_coding/codecs/vp8/include',
'modules/video_coding/codecs/vp9/include',
'modules/video_coding/include',
'rtc_base',
'system_wrappers/include',
'voice_engine/include',
)
API_DIRS = NATIVE_API_DIRS[:] + LEGACY_API_DIRS[:]
@ -331,8 +330,7 @@ def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api):
cwd = input_api.PresubmitLocalPath()
script_path = os.path.join('tools_webrtc', 'presubmit_checks_lib',
'check_package_boundaries.py')
webrtc_path = os.path.join('webrtc')
command = [sys.executable, script_path, webrtc_path]
command = [sys.executable, script_path]
command += [gn_file.LocalPath() for gn_file in gn_files]
returncode, _, stderr = _RunCommand(command, cwd)
if returncode:
@ -347,8 +345,7 @@ def CheckGnChanges(input_api, output_api):
gn_files = []
for f in input_api.AffectedSourceFiles(source_file_filter):
if f.LocalPath().startswith('webrtc'):
gn_files.append(f)
gn_files.append(f)
result = []
if gn_files:
@ -494,9 +491,9 @@ def RunPythonTests(input_api, output_api):
test_directories = [
input_api.PresubmitLocalPath(),
Join('webrtc', 'rtc_tools', 'py_event_log_analyzer'),
Join('webrtc', 'rtc_tools'),
Join('webrtc', 'audio', 'test', 'unittests'),
Join('rtc_tools', 'py_event_log_analyzer'),
Join('rtc_tools'),
Join('audio', 'test', 'unittests'),
] + [
root for root, _, files in os.walk(Join('tools_webrtc'))
if any(f.endswith('_test.py') for f in files)
@ -517,7 +514,7 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api):
"""Checks that the namespace google::protobuf has not been used."""
files = []
pattern = input_api.re.compile(r'google::protobuf')
proto_utils_path = os.path.join('webrtc', 'rtc_base', 'protobuf_utils.h')
proto_utils_path = os.path.join('rtc_base', 'protobuf_utils.h')
for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']:
continue
@ -533,6 +530,28 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api):
return []
def _LicenseHeader(input_api):
"""Returns the license header regexp."""
# Accept any year number from 2003 to the current year
current_year = int(input_api.time.strftime('%Y'))
allowed_years = (str(s) for s in reversed(xrange(2003, current_year + 1)))
years_re = '(' + '|'.join(allowed_years) + ')'
license_header = (
r'.*? Copyright( \(c\))? %(year)s The WebRTC [Pp]roject [Aa]uthors\. '
r'All [Rr]ights [Rr]eserved\.\n'
r'.*?\n'
r'.*? Use of this source code is governed by a BSD-style license\n'
r'.*? that can be found in the LICENSE file in the root of the source\n'
r'.*? tree\. An additional intellectual property rights grant can be '
r'found\n'
r'.*? in the file PATENTS\. All contributing project authors may\n'
r'.*? be found in the AUTHORS file in the root of the source tree\.\n'
) % {
'year': years_re,
}
return license_header
def CommonChecks(input_api, output_api):
"""Checks common to both upload and commit."""
results = []
@ -541,11 +560,12 @@ def CommonChecks(input_api, output_api):
black_list = input_api.DEFAULT_BLACK_LIST + (
r".*\bobjc[\\\/].*",
r".*objc\.[hcm]+$",
r"webrtc\/build\/ios\/SDK\/.*",
)
source_file_filter = lambda x: input_api.FilterSourceFile(x, None, black_list)
results.extend(CheckApprovedFilesLintClean(
input_api, output_api, source_file_filter))
results.extend(input_api.canned_checks.CheckLicense(
input_api, output_api, _LicenseHeader(input_api)))
results.extend(input_api.canned_checks.RunPylint(input_api, output_api,
black_list=(r'^base[\\\/].*\.py$',
r'^build[\\\/].*\.py$',
@ -599,8 +619,13 @@ def CommonChecks(input_api, output_api):
results.extend(CheckJSONParseErrors(input_api, output_api))
results.extend(RunPythonTests(input_api, output_api))
results.extend(CheckUsageOfGoogleProtobufNamespace(input_api, output_api))
results.extend(CheckOrphanHeaders(input_api, output_api))
results.extend(CheckNewLineAtTheEndOfProtoFiles(input_api, output_api))
# TODO(mbonadei): re-enable after the migration from src/webrtc to src/
# in order to avoid to trigger an error for each orphan header (we are
# moving all of them).
# results.extend(CheckOrphanHeaders(input_api, output_api))
# TODO(mbonadei): check before re-enable because it seems it is reporting
# some false positives.
# results.extend(CheckNewLineAtTheEndOfProtoFiles(input_api, output_api))
return results

24
api/DEPS
View file

@ -1,30 +1,30 @@
include_rules = [
"+third_party/libyuv",
"+webrtc/common_video",
"+webrtc/media",
"+webrtc/p2p",
"+webrtc/pc",
"+common_video",
"+media",
"+p2p",
"+pc",
]
specific_include_rules = {
"peerconnection_jni\.cc": [
"+webrtc/voice_engine",
"+voice_engine",
],
# TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h
# has moved to api/.
"peerconnectioninterface\.h": [
"+webrtc/call/callfactoryinterface.h",
"+webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h",
"+webrtc/modules/audio_coding/codecs/audio_encoder_factory.h",
"+webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
"+call/callfactoryinterface.h",
"+logging/rtc_event_log/rtc_event_log_factory_interface.h",
"+modules/audio_coding/codecs/audio_encoder_factory.h",
"+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
],
# Needed because AudioEncoderOpus is in the wrong place for
# backwards compatibilty reasons. See
# https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
"audio_encoder_opus\.h": [
"+webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h",
"+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
],
# We allow .cc files in webrtc/api/ to #include a bunch of stuff
@ -32,10 +32,10 @@ specific_include_rules = {
# their #includes to whoever's #including them, but .cc files do not
# since no one #includes them.
".*\.cc": [
"+webrtc/modules/audio_coding",
"+modules/audio_coding",
],
".*i420_buffer\.h": [
"+webrtc/system_wrappers/include/aligned_malloc.h",
"+system_wrappers/include/aligned_malloc.h",
],
}

10
api/array_view.h
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ARRAY_VIEW_H_
#define WEBRTC_API_ARRAY_VIEW_H_
#ifndef API_ARRAY_VIEW_H_
#define API_ARRAY_VIEW_H_
#include <algorithm>
#include <type_traits>
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/type_traits.h"
#include "rtc_base/checks.h"
#include "rtc_base/type_traits.h"
namespace rtc {
@ -260,4 +260,4 @@ inline ArrayView<T> MakeArrayView(T* data, size_t size) {
} // namespace rtc
#endif // WEBRTC_API_ARRAY_VIEW_H_
#endif // API_ARRAY_VIEW_H_

10
api/array_view_unittest.cc
View file

@ -13,11 +13,11 @@
#include <utility>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/gunit.h"
#include "webrtc/test/gmock.h"
#include "api/array_view.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "test/gmock.h"
namespace rtc {

10
api/audio/audio_mixer.h
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_AUDIO_MIXER_H_
#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_
#ifndef API_AUDIO_AUDIO_MIXER_H_
#define API_AUDIO_AUDIO_MIXER_H_
#include <memory>
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/rtc_base/refcount.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@ -77,4 +77,4 @@ class AudioMixer : public rtc::RefCountInterface {
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_AUDIO_MIXER_H_
#endif // API_AUDIO_AUDIO_MIXER_H_

12
api/audio_codecs/L16/audio_decoder_L16.cc
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
namespace webrtc {

12
api/audio_codecs/L16/audio_decoder_L16.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#ifndef API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -41,4 +41,4 @@ struct AudioDecoderL16 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#endif // API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_

12
api/audio_codecs/L16/audio_encoder_L16.cc
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
namespace webrtc {

12
api/audio_codecs/L16/audio_encoder_L16.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#ifndef API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -45,4 +45,4 @@ struct AudioEncoderL16 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#endif // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_

10
api/audio_codecs/audio_decoder.cc
View file

@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_decoder.h"
#include <assert.h>
#include <memory>
#include <utility>
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/sanitizer.h"
#include "webrtc/rtc_base/trace_event.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/sanitizer.h"
#include "rtc_base/trace_event.h"
namespace webrtc {

16
api/audio_codecs/audio_decoder.h
View file

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
#ifndef API_AUDIO_CODECS_AUDIO_DECODER_H_
#define API_AUDIO_CODECS_AUDIO_DECODER_H_
#include <memory>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/typedefs.h"
#include "api/array_view.h"
#include "api/optional.h"
#include "rtc_base/buffer.h"
#include "rtc_base/constructormagic.h"
#include "typedefs.h"
namespace webrtc {
@ -174,4 +174,4 @@ class AudioDecoder {
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_
#endif // API_AUDIO_CODECS_AUDIO_DECODER_H_

12
api/audio_codecs/audio_decoder_factory.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/rtc_base/refcount.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@ -34,4 +34,4 @@ class AudioDecoderFactory : public rtc::RefCountInterface {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#endif // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_

10
api/audio_codecs/audio_decoder_factory_template.h
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@ -121,4 +121,4 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory() {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#endif // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_

6
api/audio_codecs/audio_encoder.cc
View file

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_encoder.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/trace_event.h"
#include "rtc_base/checks.h"
#include "rtc_base/trace_event.h"
namespace webrtc {

16
api/audio_codecs/audio_encoder.h
View file

@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
#define API_AUDIO_CODECS_AUDIO_ENCODER_H_
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/api/optional.h"
#include "webrtc/rtc_base/buffer.h"
#include "webrtc/rtc_base/deprecation.h"
#include "webrtc/typedefs.h"
#include "api/array_view.h"
#include "api/optional.h"
#include "rtc_base/buffer.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h"
namespace webrtc {
@ -248,4 +248,4 @@ class AudioEncoder {
rtc::Buffer* encoded) = 0;
};
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_
#endif // API_AUDIO_CODECS_AUDIO_ENCODER_H_

12
api/audio_codecs/audio_encoder_factory.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/rtc_base/refcount.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@ -43,4 +43,4 @@ class AudioEncoderFactory : public rtc::RefCountInterface {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#endif // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_

10
api/audio_codecs/audio_encoder_factory_template.h
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@ -139,4 +139,4 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory() {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#endif // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_

4
api/audio_codecs/audio_format.cc
View file

@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/audio_format.h"
#include "api/audio_codecs/audio_format.h"
#include "webrtc/common_types.h"
#include "common_types.h"
namespace webrtc {

8
api/audio_codecs/audio_format.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <ostream>
#include <string>
#include <utility>
#include "webrtc/api/optional.h"
#include "api/optional.h"
namespace webrtc {
@ -139,4 +139,4 @@ std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs);
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_
#endif // API_AUDIO_CODECS_AUDIO_FORMAT_H_

18
api/audio_codecs/builtin_audio_decoder_factory.cc
View file

@ -8,27 +8,27 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h"
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h" // nogncheck
#include "api/audio_codecs/g722/audio_decoder_g722.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ILBC
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ISAC_FIX
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h" // nogncheck
#include "api/audio_codecs/isac/audio_decoder_isac_fix.h" // nogncheck
#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h" // nogncheck
#include "api/audio_codecs/isac/audio_decoder_isac_float.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_OPUS
#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
#endif
namespace webrtc {

10
api/audio_codecs/builtin_audio_decoder_factory.h
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#define API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@ -22,4 +22,4 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#endif // API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

18
api/audio_codecs/builtin_audio_encoder_factory.cc
View file

@ -8,27 +8,27 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h" // nogncheck
#include "api/audio_codecs/g722/audio_encoder_g722.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ILBC
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_ISAC_FIX
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h" // nogncheck
#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" // nogncheck
#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h" // nogncheck
#include "api/audio_codecs/isac/audio_encoder_isac_float.h" // nogncheck
#endif
#if WEBRTC_USE_BUILTIN_OPUS
#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
#endif
namespace webrtc {

10
api/audio_codecs/builtin_audio_encoder_factory.h
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#define API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@ -22,4 +22,4 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory();
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#endif // API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_

10
api/audio_codecs/g711/audio_decoder_g711.cc
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
namespace webrtc {

12
api/audio_codecs/g711/audio_decoder_g711.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#define WEBRTC_API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#ifndef API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -40,4 +40,4 @@ struct AudioDecoderG711 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#endif // API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_

14
api/audio_codecs/g711/audio_encoder_g711.cc
View file

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/safe_minmax.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {

12
api/audio_codecs/g711/audio_encoder_g711.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#define WEBRTC_API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#ifndef API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -45,4 +45,4 @@ struct AudioEncoderG711 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#endif // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_

10
api/audio_codecs/g722/audio_decoder_g722.cc
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h"
#include "api/audio_codecs/g722/audio_decoder_g722.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
namespace webrtc {

12
api/audio_codecs/g722/audio_decoder_g722.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#ifndef API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -36,4 +36,4 @@ struct AudioDecoderG722 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#endif // API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_

14
api/audio_codecs/g722/audio_encoder_g722.cc
View file

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/safe_minmax.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {

14
api/audio_codecs/g722/audio_encoder_g722.h
View file

@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/optional.h"
namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderG722 {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#endif // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_

6
api/audio_codecs/g722/audio_encoder_g722_config.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
#ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
namespace webrtc {
@ -24,4 +24,4 @@ struct AudioEncoderG722Config {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
#endif // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_

8
api/audio_codecs/ilbc/audio_decoder_ilbc.cc
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {

12
api/audio_codecs/ilbc/audio_decoder_ilbc.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -33,4 +33,4 @@ struct AudioDecoderIlbc {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#endif // API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_

14
api/audio_codecs/ilbc/audio_encoder_ilbc.cc
View file

@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include <memory>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/safe_conversions.h"
#include "rtc_base/safe_minmax.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {
namespace {

14
api/audio_codecs/ilbc/audio_encoder_ilbc.h
View file

@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "api/optional.h"
namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderIlbc {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_

6
api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
namespace webrtc {
@ -26,4 +26,4 @@ struct AudioEncoderIlbcConfig {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_

8
api/audio_codecs/isac/audio_decoder_isac_fix.cc
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h"
#include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {

12
api/audio_codecs/isac/audio_decoder_isac_fix.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -33,4 +33,4 @@ struct AudioDecoderIsacFix {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_

8
api/audio_codecs/isac/audio_decoder_isac_float.cc
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {

12
api/audio_codecs/isac/audio_decoder_isac_float.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -38,4 +38,4 @@ struct AudioDecoderIsacFloat {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_

10
api/audio_codecs/isac/audio_encoder_isac_fix.cc
View file

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {

12
api/audio_codecs/isac/audio_encoder_isac_fix.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -38,4 +38,4 @@ struct AudioEncoderIsacFix {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_

10
api/audio_codecs/isac/audio_encoder_isac_float.cc
View file

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/string_to_number.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/string_to_number.h"
namespace webrtc {

12
api/audio_codecs/isac/audio_encoder_isac_float.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -43,4 +43,4 @@ struct AudioEncoderIsacFloat {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_

8
api/audio_codecs/opus/audio_decoder_opus.cc
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "common_types.h"
#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {

12
api/audio_codecs/opus/audio_decoder_opus.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/optional.h"
namespace webrtc {
@ -35,4 +35,4 @@ struct AudioDecoderOpus {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#endif // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_

8
api/audio_codecs/opus/audio_encoder_opus.h
View file

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderOpus {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_

2
api/audio_codecs/opus/audio_encoder_opus_config.cc
View file

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
namespace webrtc {

8
api/audio_codecs/opus/audio_encoder_opus_config.h
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#include <stddef.h>
#include <vector>
#include "webrtc/api/optional.h"
#include "api/optional.h"
namespace webrtc {
@ -70,4 +70,4 @@ struct AudioEncoderOpusConfig {
} // namespace webrtc
#endif // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#endif // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_

24
api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
View file

@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h"
#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h"
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_audio_decoder.h"
#include "api/audio_codecs/audio_decoder_factory_template.h"
#include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "api/audio_codecs/g711/audio_decoder_g711.h"
#include "api/audio_codecs/g722/audio_decoder_g722.h"
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
#include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "rtc_base/ptr_util.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder.h"
namespace webrtc {

24
api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
View file

@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_audio_encoder.h"
#include "api/audio_codecs/audio_encoder_factory_template.h"
#include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "api/audio_codecs/g711/audio_encoder_g711.h"
#include "api/audio_codecs/g722/audio_encoder_g722.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "rtc_base/ptr_util.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
namespace webrtc {

6
api/call/audio_sink.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_CALL_AUDIO_SINK_H_
#define WEBRTC_API_CALL_AUDIO_SINK_H_
#ifndef API_CALL_AUDIO_SINK_H_
#define API_CALL_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_API_CALL_AUDIO_SINK_H_
#endif // API_CALL_AUDIO_SINK_H_

6
api/call/transport.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_CALL_TRANSPORT_H_
#define WEBRTC_API_CALL_TRANSPORT_H_
#ifndef API_CALL_TRANSPORT_H_
#define API_CALL_TRANSPORT_H_
#include <stddef.h>
#include <stdint.h>
@ -37,4 +37,4 @@ class Transport {
} // namespace webrtc
#endif // WEBRTC_API_CALL_TRANSPORT_H_
#endif // API_CALL_TRANSPORT_H_

8
api/datachannel.h
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_DATACHANNEL_H_
#define WEBRTC_API_DATACHANNEL_H_
#ifndef API_DATACHANNEL_H_
#define API_DATACHANNEL_H_
// Including this file is deprecated. It is no longer part of the public API.
// This only includes the file in its new location for backwards compatibility.
#include "webrtc/pc/datachannel.h"
#include "pc/datachannel.h"
#endif // WEBRTC_API_DATACHANNEL_H_
#endif // API_DATACHANNEL_H_

14
api/datachannelinterface.h
View file

@ -11,15 +11,15 @@
// This file contains interfaces for DataChannels
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_
#define WEBRTC_API_DATACHANNELINTERFACE_H_
#ifndef API_DATACHANNELINTERFACE_H_
#define API_DATACHANNELINTERFACE_H_
#include <string>
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/copyonwritebuffer.h"
#include "webrtc/rtc_base/refcount.h"
#include "rtc_base/basictypes.h"
#include "rtc_base/checks.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@ -180,4 +180,4 @@ class DataChannelInterface : public rtc::RefCountInterface {
} // namespace webrtc
#endif // WEBRTC_API_DATACHANNELINTERFACE_H_
#endif // API_DATACHANNELINTERFACE_H_

10
api/dtmfsenderinterface.h
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_
#define WEBRTC_API_DTMFSENDERINTERFACE_H_
#ifndef API_DTMFSENDERINTERFACE_H_
#define API_DTMFSENDERINTERFACE_H_
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/rtc_base/refcount.h"
#include "api/mediastreaminterface.h"
#include "rtc_base/refcount.h"
namespace webrtc {
@ -94,4 +94,4 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
} // namespace webrtc
#endif // WEBRTC_API_DTMFSENDERINTERFACE_H_
#endif // API_DTMFSENDERINTERFACE_H_

4
api/fakemetricsobserver.cc
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/fakemetricsobserver.h"
#include "webrtc/rtc_base/checks.h"
#include "api/fakemetricsobserver.h"
#include "rtc_base/checks.h"
namespace webrtc {

10
api/fakemetricsobserver.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_FAKEMETRICSOBSERVER_H_
#define WEBRTC_API_FAKEMETRICSOBSERVER_H_
#ifndef API_FAKEMETRICSOBSERVER_H_
#define API_FAKEMETRICSOBSERVER_H_
#include <map>
#include <string>
#include <vector>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/rtc_base/thread_checker.h"
#include "api/peerconnectioninterface.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
@ -49,4 +49,4 @@ class FakeMetricsObserver : public MetricsObserverInterface {
} // namespace webrtc
#endif // WEBRTC_API_FAKEMETRICSOBSERVER_H_
#endif // API_FAKEMETRICSOBSERVER_H_

8
api/jsep.h
View file

@ -17,15 +17,15 @@
// Though in the future, we're planning to provide an SDP parsing API, with a
// structure more friendly than cricket::SessionDescription.
#ifndef WEBRTC_API_JSEP_H_
#define WEBRTC_API_JSEP_H_
#ifndef API_JSEP_H_
#define API_JSEP_H_
#include <stddef.h>
#include <string>
#include <vector>
#include "webrtc/rtc_base/refcount.h"
#include "rtc_base/refcount.h"
namespace cricket {
class Candidate;
@ -172,4 +172,4 @@ class SetSessionDescriptionObserver : public rtc::RefCountInterface {
} // namespace webrtc
#endif // WEBRTC_API_JSEP_H_
#endif // API_JSEP_H_

12
api/jsepicecandidate.h
View file

@ -11,16 +11,16 @@
// TODO(deadbeef): Move this out of api/; it's an implementation detail and
// shouldn't be used externally.
#ifndef WEBRTC_API_JSEPICECANDIDATE_H_
#define WEBRTC_API_JSEPICECANDIDATE_H_
#ifndef API_JSEPICECANDIDATE_H_
#define API_JSEPICECANDIDATE_H_
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/jsep.h"
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/jsep.h"
#include "p2p/base/candidate.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
@ -90,4 +90,4 @@ class JsepCandidateCollection : public IceCandidateCollection {
} // namespace webrtc
#endif // WEBRTC_API_JSEPICECANDIDATE_H_
#endif // API_JSEPICECANDIDATE_H_

14
api/jsepsessiondescription.h
View file

@ -11,17 +11,17 @@
// TODO(deadbeef): Move this out of api/; it's an implementation detail and
// shouldn't be used externally.
#ifndef WEBRTC_API_JSEPSESSIONDESCRIPTION_H_
#define WEBRTC_API_JSEPSESSIONDESCRIPTION_H_
#ifndef API_JSEPSESSIONDESCRIPTION_H_
#define API_JSEPSESSIONDESCRIPTION_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/jsep.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "api/jsep.h"
#include "api/jsepicecandidate.h"
#include "p2p/base/candidate.h"
#include "rtc_base/constructormagic.h"
namespace cricket {
class SessionDescription;
@ -86,4 +86,4 @@ class JsepSessionDescription : public SessionDescriptionInterface {
} // namespace webrtc
#endif // WEBRTC_API_JSEPSESSIONDESCRIPTION_H_
#endif // API_JSEPSESSIONDESCRIPTION_H_

6
api/mediaconstraintsinterface.cc
View file

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/mediaconstraintsinterface.h"
#include "api/mediaconstraintsinterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/rtc_base/stringencode.h"
#include "api/peerconnectioninterface.h"
#include "rtc_base/stringencode.h"
namespace {

10
api/mediaconstraintsinterface.h
View file

@ -17,14 +17,14 @@
// from WebRTC too.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
#ifndef WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_
#define WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_
#ifndef API_MEDIACONSTRAINTSINTERFACE_H_
#define API_MEDIACONSTRAINTSINTERFACE_H_
#include <string>
#include <vector>
#include "webrtc/api/optional.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "api/optional.h"
#include "api/peerconnectioninterface.h"
namespace webrtc {
@ -151,4 +151,4 @@ void CopyConstraintsIntoAudioOptions(
} // namespace webrtc
#endif // WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_
#endif // API_MEDIACONSTRAINTSINTERFACE_H_

8
api/mediastream.h
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_MEDIASTREAM_H_
#define WEBRTC_API_MEDIASTREAM_H_
#ifndef API_MEDIASTREAM_H_
#define API_MEDIASTREAM_H_
// Including this file is deprecated. It is no longer part of the public API.
// This only includes the file in its new location for backwards compatibility.
#include "webrtc/pc/mediastream.h"
#include "pc/mediastream.h"
#endif // WEBRTC_API_MEDIASTREAM_H_
#endif // API_MEDIASTREAM_H_

2
api/mediastreaminterface.cc
View file

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/mediastreaminterface.h"
#include "api/mediastreaminterface.h"
namespace webrtc {

26
api/mediastreaminterface.h
View file

@ -14,27 +14,27 @@
// interfaces must be used only with PeerConnection. PeerConnectionManager
// interface provides the factory methods to create MediaStream and MediaTracks.
#ifndef WEBRTC_API_MEDIASTREAMINTERFACE_H_
#define WEBRTC_API_MEDIASTREAMINTERFACE_H_
#ifndef API_MEDIASTREAMINTERFACE_H_
#define API_MEDIASTREAMINTERFACE_H_
#include <stddef.h>
#include <string>
#include <vector>
#include "webrtc/api/optional.h"
#include "webrtc/api/video/video_frame.h"
#include "api/optional.h"
#include "api/video/video_frame.h"
// TODO(zhihuang): Remove unrelated headers once downstream applications stop
// relying on them; they were previously transitively included by
// mediachannel.h, which is no longer a dependency of this file.
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/rtc_base/ratetracker.h"
#include "webrtc/rtc_base/refcount.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread.h"
#include "webrtc/rtc_base/timeutils.h"
#include "media/base/streamparams.h"
#include "media/base/videosinkinterface.h"
#include "media/base/videosourceinterface.h"
#include "rtc_base/ratetracker.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
#include "rtc_base/timeutils.h"
namespace webrtc {
@ -321,4 +321,4 @@ class MediaStreamInterface : public rtc::RefCountInterface,
} // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_
#endif // API_MEDIASTREAMINTERFACE_H_

10
api/mediastreamproxy.h
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_MEDIASTREAMPROXY_H_
#define WEBRTC_API_MEDIASTREAMPROXY_H_
#ifndef API_MEDIASTREAMPROXY_H_
#define API_MEDIASTREAMPROXY_H_
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "api/mediastreaminterface.h"
#include "api/proxy.h"
namespace webrtc {
@ -41,4 +41,4 @@ END_PROXY_MAP()
} // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMPROXY_H_
#endif // API_MEDIASTREAMPROXY_H_

8
api/mediastreamtrack.h
View file

@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_MEDIASTREAMTRACK_H_
#define WEBRTC_API_MEDIASTREAMTRACK_H_
#ifndef API_MEDIASTREAMTRACK_H_
#define API_MEDIASTREAMTRACK_H_
// Including this file is deprecated. It is no longer part of the public API.
// This only includes the file in its new location for backwards compatibility.
#include "webrtc/pc/mediastreamtrack.h"
#include "pc/mediastreamtrack.h"
#endif // WEBRTC_API_MEDIASTREAMTRACK_H_
#endif // API_MEDIASTREAMTRACK_H_

10
api/mediastreamtrackproxy.h
View file

@ -11,13 +11,13 @@
// This file includes proxy classes for tracks. The purpose is
// to make sure tracks are only accessed from the signaling thread.
#ifndef WEBRTC_API_MEDIASTREAMTRACKPROXY_H_
#define WEBRTC_API_MEDIASTREAMTRACKPROXY_H_
#ifndef API_MEDIASTREAMTRACKPROXY_H_
#define API_MEDIASTREAMTRACKPROXY_H_
#include <string>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "api/mediastreaminterface.h"
#include "api/proxy.h"
namespace webrtc {
@ -62,4 +62,4 @@ END_PROXY_MAP()
} // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMTRACKPROXY_H_
#endif // API_MEDIASTREAMTRACKPROXY_H_

6
api/mediatypes.cc
View file

@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/mediatypes.h"
#include "api/mediatypes.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/rtc_base/checks.h"
#include "api/mediastreaminterface.h"
#include "rtc_base/checks.h"
namespace {
static const char* kMediaTypeData = "data";

6
api/mediatypes.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_MEDIATYPES_H_
#define WEBRTC_API_MEDIATYPES_H_
#ifndef API_MEDIATYPES_H_
#define API_MEDIATYPES_H_
#include <string>
@ -28,4 +28,4 @@ MediaType MediaTypeFromString(const std::string& type_str);
} // namespace cricket
#endif // WEBRTC_API_MEDIATYPES_H_
#endif // API_MEDIATYPES_H_

10
api/notifier.h
View file

@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_NOTIFIER_H_
#define WEBRTC_API_NOTIFIER_H_
#ifndef API_NOTIFIER_H_
#define API_NOTIFIER_H_
#include <list>
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/rtc_base/checks.h"
#include "api/mediastreaminterface.h"
#include "rtc_base/checks.h"
namespace webrtc {
@ -58,4 +58,4 @@ class Notifier : public T {
} // namespace webrtc
#endif // WEBRTC_API_NOTIFIER_H_
#endif // API_NOTIFIER_H_

2
api/optional.cc
View file

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/optional.h"
#include "api/optional.h"
namespace rtc {
namespace optional_internal {

12
api/optional.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_OPTIONAL_H_
#define WEBRTC_API_OPTIONAL_H_
#ifndef API_OPTIONAL_H_
#define API_OPTIONAL_H_
#include <algorithm>
#include <memory>
@ -20,9 +20,9 @@
#include <ostream>
#endif // UNIT_TEST
#include "webrtc/api/array_view.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/sanitizer.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/sanitizer.h"
namespace rtc {
@ -404,4 +404,4 @@ void PrintTo(const rtc::Optional<T>& opt, std::ostream* os) {
} // namespace rtc
#endif // WEBRTC_API_OPTIONAL_H_
#endif // API_OPTIONAL_H_

4
api/optional_unittest.cc
View file

@ -14,8 +14,8 @@
#include <utility>
#include <vector>
#include "webrtc/api/optional.h"
#include "webrtc/rtc_base/gunit.h"
#include "api/optional.h"
#include "rtc_base/gunit.h"
namespace rtc {

2
api/ortc/mediadescription.cc
View file

@ -8,6 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/ortc/mediadescription.h"
#include "api/ortc/mediadescription.h"
namespace webrtc {}

10
api/ortc/mediadescription.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_MEDIADESCRIPTION_H_
#define WEBRTC_API_ORTC_MEDIADESCRIPTION_H_
#ifndef API_ORTC_MEDIADESCRIPTION_H_
#define API_ORTC_MEDIADESCRIPTION_H_
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/optional.h"
#include "webrtc/media/base/cryptoparams.h"
#include "api/optional.h"
#include "media/base/cryptoparams.h"
namespace webrtc {
@ -50,4 +50,4 @@ class MediaDescription {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_MEDIADESCRIPTION_H_
#endif // API_ORTC_MEDIADESCRIPTION_H_

4
api/ortc/mediadescription_unittest.cc
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/ortc/mediadescription.h"
#include "webrtc/test/gtest.h"
#include "api/ortc/mediadescription.h"
#include "test/gtest.h"
namespace webrtc {

38
api/ortc/ortcfactoryinterface.h
View file

@ -8,29 +8,29 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_
#ifndef API_ORTC_ORTCFACTORYINTERFACE_H_
#define API_ORTC_ORTCFACTORYINTERFACE_H_
#include <memory>
#include <string>
#include <utility> // For std::move.
#include "webrtc/api/mediaconstraintsinterface.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/ortc/ortcrtpreceiverinterface.h"
#include "webrtc/api/ortc/ortcrtpsenderinterface.h"
#include "webrtc/api/ortc/packettransportinterface.h"
#include "webrtc/api/ortc/rtptransportcontrollerinterface.h"
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/api/ortc/srtptransportinterface.h"
#include "webrtc/api/ortc/udptransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/p2p/base/packetsocketfactory.h"
#include "webrtc/rtc_base/network.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread.h"
#include "api/mediaconstraintsinterface.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/ortc/ortcrtpreceiverinterface.h"
#include "api/ortc/ortcrtpsenderinterface.h"
#include "api/ortc/packettransportinterface.h"
#include "api/ortc/rtptransportcontrollerinterface.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/ortc/srtptransportinterface.h"
#include "api/ortc/udptransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
#include "p2p/base/packetsocketfactory.h"
#include "rtc_base/network.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
namespace webrtc {
@ -236,4 +236,4 @@ class OrtcFactoryInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_
#endif // API_ORTC_ORTCFACTORYINTERFACE_H_

16
api/ortc/ortcrtpreceiverinterface.h
View file

@ -15,14 +15,14 @@
// DtlsTransport. This is to allow different types of RTP transports (besides
// DTLS-SRTP) to be used.
#ifndef WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#ifndef API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#define API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/api/rtpparameters.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
namespace webrtc {
@ -81,4 +81,4 @@ class OrtcRtpReceiverInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#endif // API_ORTC_ORTCRTPRECEIVERINTERFACE_H_

16
api/ortc/ortcrtpsenderinterface.h
View file

@ -15,14 +15,14 @@
// DtlsTransport. This is to allow different types of RTP transports (besides
// DTLS-SRTP) to be used.
#ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
#ifndef API_ORTC_ORTCRTPSENDERINTERFACE_H_
#define API_ORTC_ORTCRTPSENDERINTERFACE_H_
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h"
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/api/rtpparameters.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
namespace webrtc {
@ -74,4 +74,4 @@ class OrtcRtpSenderInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
#endif // API_ORTC_ORTCRTPSENDERINTERFACE_H_

6
api/ortc/packettransportinterface.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_
#ifndef API_ORTC_PACKETTRANSPORTINTERFACE_H_
#define API_ORTC_PACKETTRANSPORTINTERFACE_H_
namespace rtc {
@ -35,4 +35,4 @@ class PacketTransportInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_
#endif // API_ORTC_PACKETTRANSPORTINTERFACE_H_

8
api/ortc/rtptransportcontrollerinterface.h
View file

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
#define WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
#ifndef API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
#define API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
#include <vector>
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "api/ortc/rtptransportinterface.h"
namespace webrtc {
@ -54,4 +54,4 @@ class RtpTransportControllerInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
#endif // API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_

14
api/ortc/rtptransportinterface.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
#ifndef API_ORTC_RTPTRANSPORTINTERFACE_H_
#define API_ORTC_RTPTRANSPORTINTERFACE_H_
#include <string>
#include "webrtc/api/optional.h"
#include "webrtc/api/ortc/packettransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/common_types.h"
#include "api/optional.h"
#include "api/ortc/packettransportinterface.h"
#include "api/rtcerror.h"
#include "common_types.h"
namespace webrtc {
@ -120,4 +120,4 @@ class RtpTransportInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_
#endif // API_ORTC_RTPTRANSPORTINTERFACE_H_

2
api/ortc/sessiondescription.cc
View file

@ -8,6 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/ortc/sessiondescription.h"
#include "api/ortc/sessiondescription.h"
namespace webrtc {}

6
api/ortc/sessiondescription.h
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_SESSIONDESCRIPTION_H_
#define WEBRTC_API_ORTC_SESSIONDESCRIPTION_H_
#ifndef API_ORTC_SESSIONDESCRIPTION_H_
#define API_ORTC_SESSIONDESCRIPTION_H_
#include <string>
#include <utility>
@ -42,4 +42,4 @@ class SessionDescription {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_SESSIONDESCRIPTION_H_
#endif // API_ORTC_SESSIONDESCRIPTION_H_

4
api/ortc/sessiondescription_unittest.cc
View file

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/ortc/sessiondescription.h"
#include "webrtc/test/gtest.h"
#include "api/ortc/sessiondescription.h"
#include "test/gtest.h"
namespace webrtc {

12
api/ortc/srtptransportinterface.h
View file

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_
#ifndef API_ORTC_SRTPTRANSPORTINTERFACE_H_
#define API_ORTC_SRTPTRANSPORTINTERFACE_H_
#include "webrtc/api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/media/base/cryptoparams.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/rtcerror.h"
#include "media/base/cryptoparams.h"
namespace webrtc {
@ -45,4 +45,4 @@ class SrtpTransportInterface : public RtpTransportInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_
#endif // API_ORTC_SRTPTRANSPORTINTERFACE_H_

12
api/ortc/udptransportinterface.h
View file

@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_
#ifndef API_ORTC_UDPTRANSPORTINTERFACE_H_
#define API_ORTC_UDPTRANSPORTINTERFACE_H_
#include "webrtc/api/ortc/packettransportinterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/rtc_base/socketaddress.h"
#include "api/ortc/packettransportinterface.h"
#include "api/proxy.h"
#include "rtc_base/socketaddress.h"
namespace webrtc {
@ -46,4 +46,4 @@ class UdpTransportInterface : public virtual PacketTransportInterface {
} // namespace webrtc
#endif // WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_
#endif // API_ORTC_UDPTRANSPORTINTERFACE_H_

12
api/peerconnectionfactoryproxy.h
View file

@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#ifndef API_PEERCONNECTIONFACTORYPROXY_H_
#define API_PEERCONNECTIONFACTORYPROXY_H_
#include <memory>
#include <string>
#include <utility>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/rtc_base/bind.h"
#include "api/peerconnectioninterface.h"
#include "api/proxy.h"
#include "rtc_base/bind.h"
namespace webrtc {
@ -75,4 +75,4 @@ END_PROXY_MAP()
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#endif // API_PEERCONNECTIONFACTORYPROXY_H_

52
api/peerconnectioninterface.h
View file

@ -64,37 +64,37 @@
// 7. Once a candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. Send these candidates to the remote peer.
#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
#ifndef API_PEERCONNECTIONINTERFACE_H_
#define API_PEERCONNECTIONINTERFACE_H_
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/api/datachannelinterface.h"
#include "webrtc/api/dtmfsenderinterface.h"
#include "webrtc/api/jsep.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/rtcerror.h"
#include "webrtc/api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h"
#include "webrtc/api/stats/rtcstatscollectorcallback.h"
#include "webrtc/api/statstypes.h"
#include "webrtc/api/umametrics.h"
#include "webrtc/call/callfactoryinterface.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/rtc_base/fileutils.h"
#include "webrtc/rtc_base/network.h"
#include "webrtc/rtc_base/rtccertificate.h"
#include "webrtc/rtc_base/rtccertificategenerator.h"
#include "webrtc/rtc_base/socketaddress.h"
#include "webrtc/rtc_base/sslstreamadapter.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/datachannelinterface.h"
#include "api/dtmfsenderinterface.h"
#include "api/jsep.h"
#include "api/mediastreaminterface.h"
#include "api/rtcerror.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
#include "api/stats/rtcstatscollectorcallback.h"
#include "api/statstypes.h"
#include "api/umametrics.h"
#include "call/callfactoryinterface.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/mediachannel.h"
#include "media/base/videocapturer.h"
#include "p2p/base/portallocator.h"
#include "rtc_base/fileutils.h"
#include "rtc_base/network.h"
#include "rtc_base/rtccertificate.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/sslstreamadapter.h"
namespace rtc {
class SSLIdentity;
@ -1224,4 +1224,4 @@ CreateModularPeerConnectionFactory(
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
#endif // API_PEERCONNECTIONINTERFACE_H_

10
api/peerconnectionproxy.h
View file

@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_PEERCONNECTIONPROXY_H_
#define WEBRTC_API_PEERCONNECTIONPROXY_H_
#ifndef API_PEERCONNECTIONPROXY_H_
#define API_PEERCONNECTIONPROXY_H_
#include <string>
#include <vector>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h"
#include "api/peerconnectioninterface.h"
#include "api/proxy.h"
namespace webrtc {
@ -111,4 +111,4 @@ END_PROXY_MAP()
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONPROXY_H_
#endif // API_PEERCONNECTIONPROXY_H_

10
api/proxy.h
View file

@ -49,14 +49,14 @@
// The variant defined with BEGIN_OWNED_PROXY_MAP does not use
// refcounting, and instead just takes ownership of the object being proxied.
#ifndef WEBRTC_API_PROXY_H_
#define WEBRTC_API_PROXY_H_
#ifndef API_PROXY_H_
#define API_PROXY_H_
#include <memory>
#include <utility>
#include "webrtc/rtc_base/event.h"
#include "webrtc/rtc_base/thread.h"
#include "rtc_base/event.h"
#include "rtc_base/thread.h"
namespace webrtc {
@ -579,4 +579,4 @@ class MethodCall5 : public rtc::Message,
} // namespace webrtc
#endif // WEBRTC_API_PROXY_H_
#endif // API_PROXY_H_

4
api/rtcerror.cc
View file

@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/rtcerror.h"
#include "api/rtcerror.h"
#include "webrtc/rtc_base/arraysize.h"
#include "rtc_base/arraysize.h"
namespace {

10
api/rtcerror.h
View file

@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_RTCERROR_H_
#define WEBRTC_API_RTCERROR_H_
#ifndef API_RTCERROR_H_
#define API_RTCERROR_H_
#include <ostream>
#include <string>
#include <utility> // For std::move.
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
@ -297,4 +297,4 @@ class RTCErrorOr {
} // namespace webrtc
#endif // WEBRTC_API_RTCERROR_H_
#endif // API_RTCERROR_H_

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