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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
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

8
.gitignore vendored
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@ -57,9 +57,9 @@
/tools_webrtc/video_quality_toolchain/mac/zxing /tools_webrtc/video_quality_toolchain/mac/zxing
/tools_webrtc/video_quality_toolchain/win/*.dll /tools_webrtc/video_quality_toolchain/win/*.dll
/tools_webrtc/video_quality_toolchain/win/*.exe /tools_webrtc/video_quality_toolchain/win/*.exe
/webrtc/rtc_tools/testing/*.zip /rtc_tools/testing/*.zip
/webrtc/rtc_tools/testing/*.gz /rtc_tools/testing/*.gz
/webrtc/rtc_tools/testing/golang/*/*.gz /rtc_tools/testing/golang/*/*.gz
/webrtc/rtc_tools/testing/golang/*/*.zip /rtc_tools/testing/golang/*/*.zip
/x86-generic_out/ /x86-generic_out/
/xcodebuild /xcodebuild

24
.gn
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@ -20,7 +20,29 @@ secondary_source = "//build/secondary/"
# matching these patterns (see "gn help label_pattern" for format) will have # matching these patterns (see "gn help label_pattern" for format) will have
# their includes checked for proper dependencies when you run either # their includes checked for proper dependencies when you run either
# "gn check" or "gn gen --check". # "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 # 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 # to force additional review for new uses of exec_script, which is strongly

30
BUILD.gn
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@ -431,8 +431,8 @@ if (rtc_include_tests) {
# TODO(pbos): Rename test suite, this is no longer "just" for video targets. # TODO(pbos): Rename test suite, this is no longer "just" for video targets.
video_engine_tests_resources = [ video_engine_tests_resources = [
"../resources/foreman_cif_short.yuv", "resources/foreman_cif_short.yuv",
"../resources/voice_engine/audio_long16.pcm", "resources/voice_engine/audio_long16.pcm",
] ]
if (is_ios) { if (is_ios) {
@ -475,19 +475,19 @@ if (rtc_include_tests) {
} }
webrtc_perf_tests_resources = [ webrtc_perf_tests_resources = [
"../resources/audio_coding/speech_mono_16kHz.pcm", "resources/audio_coding/speech_mono_16kHz.pcm",
"../resources/audio_coding/speech_mono_32_48kHz.pcm", "resources/audio_coding/speech_mono_32_48kHz.pcm",
"../resources/audio_coding/testfile32kHz.pcm", "resources/audio_coding/testfile32kHz.pcm",
"../resources/ConferenceMotion_1280_720_50.yuv", "resources/ConferenceMotion_1280_720_50.yuv",
"../resources/difficult_photo_1850_1110.yuv", "resources/difficult_photo_1850_1110.yuv",
"../resources/foreman_cif.yuv", "resources/foreman_cif.yuv",
"../resources/google-wifi-3mbps.rx", "resources/google-wifi-3mbps.rx",
"../resources/paris_qcif.yuv", "resources/paris_qcif.yuv",
"../resources/photo_1850_1110.yuv", "resources/photo_1850_1110.yuv",
"../resources/presentation_1850_1110.yuv", "resources/presentation_1850_1110.yuv",
"../resources/verizon4g-downlink.rx", "resources/verizon4g-downlink.rx",
"../resources/voice_engine/audio_long16.pcm", "resources/voice_engine/audio_long16.pcm",
"../resources/web_screenshot_1850_1110.yuv", "resources/web_screenshot_1850_1110.yuv",
] ]
if (is_ios) { if (is_ios) {

26
DEPS
View File

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

137
PRESUBMIT.py
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@ -15,29 +15,29 @@ import sys
# Files and directories that are *skipped* by cpplint in the presubmit script. # Files and directories that are *skipped* by cpplint in the presubmit script.
CPPLINT_BLACKLIST = [ 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', 'tools_webrtc',
'webrtc/api/video_codecs/video_decoder.h', 'voice_engine',
'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',
] ]
# These filters will always be removed, even if the caller specifies a filter # 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). # webrtc-users@google.com (internal list).
# 4. (later) The deprecated APIs are removed. # 4. (later) The deprecated APIs are removed.
NATIVE_API_DIRS = ( NATIVE_API_DIRS = (
'webrtc', 'api',
'webrtc/api', 'media',
'webrtc/media', 'modules/audio_device/include',
'webrtc/modules/audio_device/include', 'pc',
'webrtc/pc',
) )
# These directories should not be used but are maintained only to avoid breaking # These directories should not be used but are maintained only to avoid breaking
# some legacy downstream code. # some legacy downstream code.
LEGACY_API_DIRS = ( LEGACY_API_DIRS = (
'webrtc/common_audio/include', 'common_audio/include',
'webrtc/modules/audio_coding/include', 'modules/audio_coding/include',
'webrtc/modules/audio_conference_mixer/include', 'modules/audio_conference_mixer/include',
'webrtc/modules/audio_processing/include', 'modules/audio_processing/include',
'webrtc/modules/bitrate_controller/include', 'modules/bitrate_controller/include',
'webrtc/modules/congestion_controller/include', 'modules/congestion_controller/include',
'webrtc/modules/include', 'modules/include',
'webrtc/modules/remote_bitrate_estimator/include', 'modules/remote_bitrate_estimator/include',
'webrtc/modules/rtp_rtcp/include', 'modules/rtp_rtcp/include',
'webrtc/modules/rtp_rtcp/source', 'modules/rtp_rtcp/source',
'webrtc/modules/utility/include', 'modules/utility/include',
'webrtc/modules/video_coding/codecs/h264/include', 'modules/video_coding/codecs/h264/include',
'webrtc/modules/video_coding/codecs/i420/include', 'modules/video_coding/codecs/i420/include',
'webrtc/modules/video_coding/codecs/vp8/include', 'modules/video_coding/codecs/vp8/include',
'webrtc/modules/video_coding/codecs/vp9/include', 'modules/video_coding/codecs/vp9/include',
'webrtc/modules/video_coding/include', 'modules/video_coding/include',
'webrtc/rtc_base', 'rtc_base',
'webrtc/system_wrappers/include', 'system_wrappers/include',
'webrtc/voice_engine/include', 'voice_engine/include',
) )
API_DIRS = NATIVE_API_DIRS[:] + LEGACY_API_DIRS[:] API_DIRS = NATIVE_API_DIRS[:] + LEGACY_API_DIRS[:]
@ -331,8 +330,7 @@ def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api):
cwd = input_api.PresubmitLocalPath() cwd = input_api.PresubmitLocalPath()
script_path = os.path.join('tools_webrtc', 'presubmit_checks_lib', script_path = os.path.join('tools_webrtc', 'presubmit_checks_lib',
'check_package_boundaries.py') 'check_package_boundaries.py')
webrtc_path = os.path.join('webrtc') command = [sys.executable, script_path]
command = [sys.executable, script_path, webrtc_path]
command += [gn_file.LocalPath() for gn_file in gn_files] command += [gn_file.LocalPath() for gn_file in gn_files]
returncode, _, stderr = _RunCommand(command, cwd) returncode, _, stderr = _RunCommand(command, cwd)
if returncode: if returncode:
@ -347,7 +345,6 @@ def CheckGnChanges(input_api, output_api):
gn_files = [] gn_files = []
for f in input_api.AffectedSourceFiles(source_file_filter): for f in input_api.AffectedSourceFiles(source_file_filter):
if f.LocalPath().startswith('webrtc'):
gn_files.append(f) gn_files.append(f)
result = [] result = []
@ -494,9 +491,9 @@ def RunPythonTests(input_api, output_api):
test_directories = [ test_directories = [
input_api.PresubmitLocalPath(), input_api.PresubmitLocalPath(),
Join('webrtc', 'rtc_tools', 'py_event_log_analyzer'), Join('rtc_tools', 'py_event_log_analyzer'),
Join('webrtc', 'rtc_tools'), Join('rtc_tools'),
Join('webrtc', 'audio', 'test', 'unittests'), Join('audio', 'test', 'unittests'),
] + [ ] + [
root for root, _, files in os.walk(Join('tools_webrtc')) root for root, _, files in os.walk(Join('tools_webrtc'))
if any(f.endswith('_test.py') for f in files) 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.""" """Checks that the namespace google::protobuf has not been used."""
files = [] files = []
pattern = input_api.re.compile(r'google::protobuf') 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): for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']: if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']:
continue continue
@ -533,6 +530,28 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api):
return [] 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): def CommonChecks(input_api, output_api):
"""Checks common to both upload and commit.""" """Checks common to both upload and commit."""
results = [] results = []
@ -541,11 +560,12 @@ def CommonChecks(input_api, output_api):
black_list = input_api.DEFAULT_BLACK_LIST + ( black_list = input_api.DEFAULT_BLACK_LIST + (
r".*\bobjc[\\\/].*", r".*\bobjc[\\\/].*",
r".*objc\.[hcm]+$", r".*objc\.[hcm]+$",
r"webrtc\/build\/ios\/SDK\/.*",
) )
source_file_filter = lambda x: input_api.FilterSourceFile(x, None, black_list) source_file_filter = lambda x: input_api.FilterSourceFile(x, None, black_list)
results.extend(CheckApprovedFilesLintClean( results.extend(CheckApprovedFilesLintClean(
input_api, output_api, source_file_filter)) 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, results.extend(input_api.canned_checks.RunPylint(input_api, output_api,
black_list=(r'^base[\\\/].*\.py$', black_list=(r'^base[\\\/].*\.py$',
r'^build[\\\/].*\.py$', r'^build[\\\/].*\.py$',
@ -599,8 +619,13 @@ def CommonChecks(input_api, output_api):
results.extend(CheckJSONParseErrors(input_api, output_api)) results.extend(CheckJSONParseErrors(input_api, output_api))
results.extend(RunPythonTests(input_api, output_api)) results.extend(RunPythonTests(input_api, output_api))
results.extend(CheckUsageOfGoogleProtobufNamespace(input_api, output_api)) results.extend(CheckUsageOfGoogleProtobufNamespace(input_api, output_api))
results.extend(CheckOrphanHeaders(input_api, output_api)) # TODO(mbonadei): re-enable after the migration from src/webrtc to src/
results.extend(CheckNewLineAtTheEndOfProtoFiles(input_api, output_api)) # 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 return results

24
api/DEPS
View File

@ -1,30 +1,30 @@
include_rules = [ include_rules = [
"+third_party/libyuv", "+third_party/libyuv",
"+webrtc/common_video", "+common_video",
"+webrtc/media", "+media",
"+webrtc/p2p", "+p2p",
"+webrtc/pc", "+pc",
] ]
specific_include_rules = { specific_include_rules = {
"peerconnection_jni\.cc": [ "peerconnection_jni\.cc": [
"+webrtc/voice_engine", "+voice_engine",
], ],
# TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h # TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h
# has moved to api/. # has moved to api/.
"peerconnectioninterface\.h": [ "peerconnectioninterface\.h": [
"+webrtc/call/callfactoryinterface.h", "+call/callfactoryinterface.h",
"+webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h", "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
"+webrtc/modules/audio_coding/codecs/audio_encoder_factory.h", "+modules/audio_coding/codecs/audio_encoder_factory.h",
"+webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h", "+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
], ],
# Needed because AudioEncoderOpus is in the wrong place for # Needed because AudioEncoderOpus is in the wrong place for
# backwards compatibilty reasons. See # backwards compatibilty reasons. See
# https://bugs.chromium.org/p/webrtc/issues/detail?id=7847 # https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
"audio_encoder_opus\.h": [ "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 # 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 # their #includes to whoever's #including them, but .cc files do not
# since no one #includes them. # since no one #includes them.
".*\.cc": [ ".*\.cc": [
"+webrtc/modules/audio_coding", "+modules/audio_coding",
], ],
".*i420_buffer\.h": [ ".*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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ARRAY_VIEW_H_ #ifndef API_ARRAY_VIEW_H_
#define WEBRTC_API_ARRAY_VIEW_H_ #define API_ARRAY_VIEW_H_
#include <algorithm> #include <algorithm>
#include <type_traits> #include <type_traits>
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
#include "webrtc/rtc_base/type_traits.h" #include "rtc_base/type_traits.h"
namespace rtc { namespace rtc {
@ -260,4 +260,4 @@ inline ArrayView<T> MakeArrayView(T* data, size_t size) {
} // namespace rtc } // 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 <utility>
#include <vector> #include <vector>
#include "webrtc/api/array_view.h" #include "api/array_view.h"
#include "webrtc/rtc_base/buffer.h" #include "rtc_base/buffer.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
#include "webrtc/rtc_base/gunit.h" #include "rtc_base/gunit.h"
#include "webrtc/test/gmock.h" #include "test/gmock.h"
namespace rtc { 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_AUDIO_MIXER_H_ #ifndef API_AUDIO_AUDIO_MIXER_H_
#define WEBRTC_API_AUDIO_AUDIO_MIXER_H_ #define API_AUDIO_AUDIO_MIXER_H_
#include <memory> #include <memory>
#include "webrtc/modules/include/module_common_types.h" #include "modules/include/module_common_types.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace webrtc { namespace webrtc {
@ -77,4 +77,4 @@ class AudioMixer : public rtc::RefCountInterface {
}; };
} // namespace webrtc } // 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. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
namespace webrtc { 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_ #ifndef API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_ #define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -41,4 +41,4 @@ struct AudioDecoderL16 {
} // namespace webrtc } // 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
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h" #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/L16/audio_encoder_L16.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ #ifndef API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_ #define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -45,4 +45,4 @@ struct AudioEncoderL16 {
} // namespace webrtc } // 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
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <assert.h>
#include <memory> #include <memory>
#include <utility> #include <utility>
#include "webrtc/api/array_view.h" #include "api/array_view.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
#include "webrtc/rtc_base/sanitizer.h" #include "rtc_base/sanitizer.h"
#include "webrtc/rtc_base/trace_event.h" #include "rtc_base/trace_event.h"
namespace webrtc { namespace webrtc {

16
api/audio_codecs/audio_decoder.h
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@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_ #ifndef API_AUDIO_CODECS_AUDIO_DECODER_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_H_ #define API_AUDIO_CODECS_AUDIO_DECODER_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/array_view.h" #include "api/array_view.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/rtc_base/buffer.h" #include "rtc_base/buffer.h"
#include "webrtc/rtc_base/constructormagic.h" #include "rtc_base/constructormagic.h"
#include "webrtc/typedefs.h" #include "typedefs.h"
namespace webrtc { namespace webrtc {
@ -174,4 +174,4 @@ class AudioDecoder {
}; };
} // namespace webrtc } // 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
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_ #ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_ #define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace webrtc { namespace webrtc {
@ -34,4 +34,4 @@ class AudioDecoderFactory : public rtc::RefCountInterface {
} // namespace webrtc } // 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
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_ #ifndef API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_ #define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
namespace webrtc { namespace webrtc {
@ -121,4 +121,4 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory() {
} // namespace webrtc } // 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
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "rtc_base/checks.h"
#include "webrtc/rtc_base/trace_event.h" #include "rtc_base/trace_event.h"
namespace webrtc { namespace webrtc {

16
api/audio_codecs/audio_encoder.h
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@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_ #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_H_ #define API_AUDIO_CODECS_AUDIO_ENCODER_H_
#include <algorithm> #include <algorithm>
#include <memory> #include <memory>
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/array_view.h" #include "api/array_view.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/rtc_base/buffer.h" #include "rtc_base/buffer.h"
#include "webrtc/rtc_base/deprecation.h" #include "rtc_base/deprecation.h"
#include "webrtc/typedefs.h" #include "typedefs.h"
namespace webrtc { namespace webrtc {
@ -248,4 +248,4 @@ class AudioEncoder {
rtc::Buffer* encoded) = 0; rtc::Buffer* encoded) = 0;
}; };
} // namespace webrtc } // 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
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_ #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_ #define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace webrtc { namespace webrtc {
@ -43,4 +43,4 @@ class AudioEncoderFactory : public rtc::RefCountInterface {
} // namespace webrtc } // 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
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_ #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_ #define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
namespace webrtc { namespace webrtc {
@ -139,4 +139,4 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory() {
} // namespace webrtc } // 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
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@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree. * 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 { namespace webrtc {

8
api/audio_codecs/audio_format.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ #ifndef API_AUDIO_CODECS_AUDIO_FORMAT_H_
#define WEBRTC_API_AUDIO_CODECS_AUDIO_FORMAT_H_ #define API_AUDIO_CODECS_AUDIO_FORMAT_H_
#include <map> #include <map>
#include <ostream> #include <ostream>
#include <string> #include <string>
#include <utility> #include <utility>
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -139,4 +139,4 @@ std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs);
} // namespace webrtc } // 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
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@ -8,27 +8,27 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h" #include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h" #include "api/audio_codecs/audio_decoder_factory_template.h"
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h" #include "api/audio_codecs/g711/audio_decoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722 #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 #endif
#if WEBRTC_USE_BUILTIN_ILBC #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 #endif
#if WEBRTC_USE_BUILTIN_ISAC_FIX #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 #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 #endif
#if WEBRTC_USE_BUILTIN_OPUS #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 #endif
namespace webrtc { namespace webrtc {

10
api/audio_codecs/builtin_audio_decoder_factory.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_ #ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_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 "api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
namespace webrtc { namespace webrtc {
@ -22,4 +22,4 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
} // namespace webrtc } // 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
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@ -8,27 +8,27 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h" #include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/audio_encoder_factory_template.h"
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g711/audio_encoder_g711.h"
#if WEBRTC_USE_BUILTIN_G722 #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 #endif
#if WEBRTC_USE_BUILTIN_ILBC #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 #endif
#if WEBRTC_USE_BUILTIN_ISAC_FIX #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 #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 #endif
#if WEBRTC_USE_BUILTIN_OPUS #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 #endif
namespace webrtc { namespace webrtc {

10
api/audio_codecs/builtin_audio_encoder_factory.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_ #ifndef API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
#define WEBRTC_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 "api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
namespace webrtc { namespace webrtc {
@ -22,4 +22,4 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory();
} // namespace webrtc } // 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
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/g711/audio_decoder_g711.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_ #ifndef API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#define WEBRTC_API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_ #define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -40,4 +40,4 @@ struct AudioDecoderG711 {
} // namespace webrtc } // 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
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@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h" #include "rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h" #include "rtc_base/string_to_number.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/g711/audio_encoder_g711.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ #ifndef API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#define WEBRTC_API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_ #define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -45,4 +45,4 @@ struct AudioEncoderG711 {
} // namespace webrtc } // 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
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h" #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/g722/audio_decoder_g722.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_ #ifndef API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_ #define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -36,4 +36,4 @@ struct AudioDecoderG722 {
} // namespace webrtc } // 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
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@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h" #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h" #include "rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h" #include "rtc_base/string_to_number.h"
namespace webrtc { namespace webrtc {

14
api/audio_codecs/g722/audio_encoder_g722.h
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_ #ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_ #define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h" #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderG722 {
} // namespace webrtc } // 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
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ #ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_ #define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
namespace webrtc { namespace webrtc {
@ -24,4 +24,4 @@ struct AudioEncoderG722Config {
} // namespace webrtc } // 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
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/ilbc/audio_decoder_ilbc.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_ #ifndef API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_ #define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -33,4 +33,4 @@ struct AudioDecoderIlbc {
} // namespace webrtc } // 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
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@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/safe_conversions.h" #include "rtc_base/safe_conversions.h"
#include "webrtc/rtc_base/safe_minmax.h" #include "rtc_base/safe_minmax.h"
#include "webrtc/rtc_base/string_to_number.h" #include "rtc_base/string_to_number.h"
namespace webrtc { namespace webrtc {
namespace { namespace {

14
api/audio_codecs/ilbc/audio_encoder_ilbc.h
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ #ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_ #define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderIlbc {
} // namespace webrtc } // 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
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_ #ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_ #define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
namespace webrtc { namespace webrtc {
@ -26,4 +26,4 @@ struct AudioEncoderIlbcConfig {
} // namespace webrtc } // 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
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" #include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/isac/audio_decoder_isac_fix.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_ #ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_ #define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -33,4 +33,4 @@ struct AudioDecoderIsacFix {
} // namespace webrtc } // 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
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" #include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/isac/audio_decoder_isac_float.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_ #ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_ #define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -38,4 +38,4 @@ struct AudioDecoderIsacFloat {
} // namespace webrtc } // 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
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/string_to_number.h" #include "rtc_base/string_to_number.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/isac/audio_encoder_isac_fix.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_ #ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_ #define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -38,4 +38,4 @@ struct AudioEncoderIsacFix {
} // namespace webrtc } // 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
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@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "common_types.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/rtc_base/string_to_number.h" #include "rtc_base/string_to_number.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/isac/audio_encoder_isac_float.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ #ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#define WEBRTC_API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ #define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -43,4 +43,4 @@ struct AudioEncoderIsacFloat {
} // namespace webrtc } // 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
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * 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 <memory>
#include <utility> #include <utility>
#include <vector> #include <vector>
#include "webrtc/common_types.h" #include "common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h" #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
namespace webrtc { namespace webrtc {

12
api/audio_codecs/opus/audio_decoder_opus.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_ #ifndef API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_ #define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#include <memory> #include <memory>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h" #include "api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h" #include "api/audio_codecs/audio_format.h"
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -35,4 +35,4 @@ struct AudioDecoderOpus {
} // namespace webrtc } // 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
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_ #ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#define WEBRTC_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 { namespace webrtc {
@ -37,4 +37,4 @@ struct AudioEncoderOpus {
} // namespace webrtc } // 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
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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 { namespace webrtc {

8
api/audio_codecs/opus/audio_encoder_opus_config.h
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@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ #ifndef API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_ #define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
#include <stddef.h> #include <stddef.h>
#include <vector> #include <vector>
#include "webrtc/api/optional.h" #include "api/optional.h"
namespace webrtc { namespace webrtc {
@ -70,4 +70,4 @@ struct AudioEncoderOpusConfig {
} // namespace webrtc } // 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
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@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h" #include "api/audio_codecs/audio_decoder_factory_template.h"
#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h" #include "api/audio_codecs/L16/audio_decoder_L16.h"
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h" #include "api/audio_codecs/g711/audio_decoder_g711.h"
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h" #include "api/audio_codecs/g722/audio_decoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h" #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h" #include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
#include "webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h" #include "api/audio_codecs/isac/audio_decoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h" #include "test/gmock.h"
#include "webrtc/test/gtest.h" #include "test/gtest.h"
#include "webrtc/test/mock_audio_decoder.h" #include "test/mock_audio_decoder.h"
namespace webrtc { namespace webrtc {

24
api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
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@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/audio_encoder_factory_template.h"
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h" #include "api/audio_codecs/L16/audio_encoder_L16.h"
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g711/audio_encoder_g711.h"
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h" #include "api/audio_codecs/g722/audio_encoder_g722.h"
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h" #include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h" #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "webrtc/rtc_base/ptr_util.h" #include "rtc_base/ptr_util.h"
#include "webrtc/test/gmock.h" #include "test/gmock.h"
#include "webrtc/test/gtest.h" #include "test/gtest.h"
#include "webrtc/test/mock_audio_encoder.h" #include "test/mock_audio_encoder.h"
namespace webrtc { namespace webrtc {

6
api/call/audio_sink.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_CALL_AUDIO_SINK_H_ #ifndef API_CALL_AUDIO_SINK_H_
#define WEBRTC_API_CALL_AUDIO_SINK_H_ #define API_CALL_AUDIO_SINK_H_
#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS) #if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
// Avoid conflict with format_macros.h. // Avoid conflict with format_macros.h.
@ -50,4 +50,4 @@ class AudioSinkInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_CALL_AUDIO_SINK_H_ #endif // API_CALL_AUDIO_SINK_H_

6
api/call/transport.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_CALL_TRANSPORT_H_ #ifndef API_CALL_TRANSPORT_H_
#define WEBRTC_API_CALL_TRANSPORT_H_ #define API_CALL_TRANSPORT_H_
#include <stddef.h> #include <stddef.h>
#include <stdint.h> #include <stdint.h>
@ -37,4 +37,4 @@ class Transport {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_CALL_TRANSPORT_H_ #endif // API_CALL_TRANSPORT_H_

8
api/datachannel.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_DATACHANNEL_H_ #ifndef API_DATACHANNEL_H_
#define WEBRTC_API_DATACHANNEL_H_ #define API_DATACHANNEL_H_
// Including this file is deprecated. It is no longer part of the public API. // 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. // 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
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@ -11,15 +11,15 @@
// This file contains interfaces for DataChannels // This file contains interfaces for DataChannels
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_ #ifndef API_DATACHANNELINTERFACE_H_
#define WEBRTC_API_DATACHANNELINTERFACE_H_ #define API_DATACHANNELINTERFACE_H_
#include <string> #include <string>
#include "webrtc/rtc_base/basictypes.h" #include "rtc_base/basictypes.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
#include "webrtc/rtc_base/copyonwritebuffer.h" #include "rtc_base/copyonwritebuffer.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace webrtc { namespace webrtc {
@ -180,4 +180,4 @@ class DataChannelInterface : public rtc::RefCountInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_DATACHANNELINTERFACE_H_ #endif // API_DATACHANNELINTERFACE_H_

10
api/dtmfsenderinterface.h
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_ #ifndef API_DTMFSENDERINTERFACE_H_
#define WEBRTC_API_DTMFSENDERINTERFACE_H_ #define API_DTMFSENDERINTERFACE_H_
#include <string> #include <string>
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace webrtc { namespace webrtc {
@ -94,4 +94,4 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_DTMFSENDERINTERFACE_H_ #endif // API_DTMFSENDERINTERFACE_H_

4
api/fakemetricsobserver.cc
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/fakemetricsobserver.h" #include "api/fakemetricsobserver.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
namespace webrtc { namespace webrtc {

10
api/fakemetricsobserver.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_FAKEMETRICSOBSERVER_H_ #ifndef API_FAKEMETRICSOBSERVER_H_
#define WEBRTC_API_FAKEMETRICSOBSERVER_H_ #define API_FAKEMETRICSOBSERVER_H_
#include <map> #include <map>
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/peerconnectioninterface.h" #include "api/peerconnectioninterface.h"
#include "webrtc/rtc_base/thread_checker.h" #include "rtc_base/thread_checker.h"
namespace webrtc { namespace webrtc {
@ -49,4 +49,4 @@ class FakeMetricsObserver : public MetricsObserverInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_FAKEMETRICSOBSERVER_H_ #endif // API_FAKEMETRICSOBSERVER_H_

8
api/jsep.h
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@ -17,15 +17,15 @@
// Though in the future, we're planning to provide an SDP parsing API, with a // Though in the future, we're planning to provide an SDP parsing API, with a
// structure more friendly than cricket::SessionDescription. // structure more friendly than cricket::SessionDescription.
#ifndef WEBRTC_API_JSEP_H_ #ifndef API_JSEP_H_
#define WEBRTC_API_JSEP_H_ #define API_JSEP_H_
#include <stddef.h> #include <stddef.h>
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
namespace cricket { namespace cricket {
class Candidate; class Candidate;
@ -172,4 +172,4 @@ class SetSessionDescriptionObserver : public rtc::RefCountInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_JSEP_H_ #endif // API_JSEP_H_

12
api/jsepicecandidate.h
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@ -11,16 +11,16 @@
// TODO(deadbeef): Move this out of api/; it's an implementation detail and // TODO(deadbeef): Move this out of api/; it's an implementation detail and
// shouldn't be used externally. // shouldn't be used externally.
#ifndef WEBRTC_API_JSEPICECANDIDATE_H_ #ifndef API_JSEPICECANDIDATE_H_
#define WEBRTC_API_JSEPICECANDIDATE_H_ #define API_JSEPICECANDIDATE_H_
#include <string> #include <string>
#include <utility> #include <utility>
#include <vector> #include <vector>
#include "webrtc/api/jsep.h" #include "api/jsep.h"
#include "webrtc/p2p/base/candidate.h" #include "p2p/base/candidate.h"
#include "webrtc/rtc_base/constructormagic.h" #include "rtc_base/constructormagic.h"
namespace webrtc { namespace webrtc {
@ -90,4 +90,4 @@ class JsepCandidateCollection : public IceCandidateCollection {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_JSEPICECANDIDATE_H_ #endif // API_JSEPICECANDIDATE_H_

14
api/jsepsessiondescription.h
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@ -11,17 +11,17 @@
// TODO(deadbeef): Move this out of api/; it's an implementation detail and // TODO(deadbeef): Move this out of api/; it's an implementation detail and
// shouldn't be used externally. // shouldn't be used externally.
#ifndef WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ #ifndef API_JSEPSESSIONDESCRIPTION_H_
#define WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ #define API_JSEPSESSIONDESCRIPTION_H_
#include <memory> #include <memory>
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/jsep.h" #include "api/jsep.h"
#include "webrtc/api/jsepicecandidate.h" #include "api/jsepicecandidate.h"
#include "webrtc/p2p/base/candidate.h" #include "p2p/base/candidate.h"
#include "webrtc/rtc_base/constructormagic.h" #include "rtc_base/constructormagic.h"
namespace cricket { namespace cricket {
class SessionDescription; class SessionDescription;
@ -86,4 +86,4 @@ class JsepSessionDescription : public SessionDescriptionInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_JSEPSESSIONDESCRIPTION_H_ #endif // API_JSEPSESSIONDESCRIPTION_H_

6
api/mediaconstraintsinterface.cc
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "api/peerconnectioninterface.h"
#include "webrtc/rtc_base/stringencode.h" #include "rtc_base/stringencode.h"
namespace { namespace {

10
api/mediaconstraintsinterface.h
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@ -17,14 +17,14 @@
// from WebRTC too. // from WebRTC too.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617 // https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
#ifndef WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ #ifndef API_MEDIACONSTRAINTSINTERFACE_H_
#define WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ #define API_MEDIACONSTRAINTSINTERFACE_H_
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/api/peerconnectioninterface.h" #include "api/peerconnectioninterface.h"
namespace webrtc { namespace webrtc {
@ -151,4 +151,4 @@ void CopyConstraintsIntoAudioOptions(
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_MEDIACONSTRAINTSINTERFACE_H_ #endif // API_MEDIACONSTRAINTSINTERFACE_H_

8
api/mediastream.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_MEDIASTREAM_H_ #ifndef API_MEDIASTREAM_H_
#define WEBRTC_API_MEDIASTREAM_H_ #define API_MEDIASTREAM_H_
// Including this file is deprecated. It is no longer part of the public API. // 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. // 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
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
namespace webrtc { namespace webrtc {

26
api/mediastreaminterface.h
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@ -14,27 +14,27 @@
// interfaces must be used only with PeerConnection. PeerConnectionManager // interfaces must be used only with PeerConnection. PeerConnectionManager
// interface provides the factory methods to create MediaStream and MediaTracks. // interface provides the factory methods to create MediaStream and MediaTracks.
#ifndef WEBRTC_API_MEDIASTREAMINTERFACE_H_ #ifndef API_MEDIASTREAMINTERFACE_H_
#define WEBRTC_API_MEDIASTREAMINTERFACE_H_ #define API_MEDIASTREAMINTERFACE_H_
#include <stddef.h> #include <stddef.h>
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/api/video/video_frame.h" #include "api/video/video_frame.h"
// TODO(zhihuang): Remove unrelated headers once downstream applications stop // TODO(zhihuang): Remove unrelated headers once downstream applications stop
// relying on them; they were previously transitively included by // relying on them; they were previously transitively included by
// mediachannel.h, which is no longer a dependency of this file. // mediachannel.h, which is no longer a dependency of this file.
#include "webrtc/media/base/streamparams.h" #include "media/base/streamparams.h"
#include "webrtc/media/base/videosinkinterface.h" #include "media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h" #include "media/base/videosourceinterface.h"
#include "webrtc/rtc_base/ratetracker.h" #include "rtc_base/ratetracker.h"
#include "webrtc/rtc_base/refcount.h" #include "rtc_base/refcount.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread.h" #include "rtc_base/thread.h"
#include "webrtc/rtc_base/timeutils.h" #include "rtc_base/timeutils.h"
namespace webrtc { namespace webrtc {
@ -321,4 +321,4 @@ class MediaStreamInterface : public rtc::RefCountInterface,
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMINTERFACE_H_ #endif // API_MEDIASTREAMINTERFACE_H_

10
api/mediastreamproxy.h
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_MEDIASTREAMPROXY_H_ #ifndef API_MEDIASTREAMPROXY_H_
#define WEBRTC_API_MEDIASTREAMPROXY_H_ #define API_MEDIASTREAMPROXY_H_
#include <string> #include <string>
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/proxy.h" #include "api/proxy.h"
namespace webrtc { namespace webrtc {
@ -41,4 +41,4 @@ END_PROXY_MAP()
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMPROXY_H_ #endif // API_MEDIASTREAMPROXY_H_

8
api/mediastreamtrack.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_MEDIASTREAMTRACK_H_ #ifndef API_MEDIASTREAMTRACK_H_
#define WEBRTC_API_MEDIASTREAMTRACK_H_ #define API_MEDIASTREAMTRACK_H_
// Including this file is deprecated. It is no longer part of the public API. // 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. // 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
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@ -11,13 +11,13 @@
// This file includes proxy classes for tracks. The purpose is // This file includes proxy classes for tracks. The purpose is
// to make sure tracks are only accessed from the signaling thread. // to make sure tracks are only accessed from the signaling thread.
#ifndef WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ #ifndef API_MEDIASTREAMTRACKPROXY_H_
#define WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ #define API_MEDIASTREAMTRACKPROXY_H_
#include <string> #include <string>
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/proxy.h" #include "api/proxy.h"
namespace webrtc { namespace webrtc {
@ -62,4 +62,4 @@ END_PROXY_MAP()
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_MEDIASTREAMTRACKPROXY_H_ #endif // API_MEDIASTREAMTRACKPROXY_H_

6
api/mediatypes.cc
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@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree. * 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 "api/mediastreaminterface.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
namespace { namespace {
static const char* kMediaTypeData = "data"; static const char* kMediaTypeData = "data";

6
api/mediatypes.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_MEDIATYPES_H_ #ifndef API_MEDIATYPES_H_
#define WEBRTC_API_MEDIATYPES_H_ #define API_MEDIATYPES_H_
#include <string> #include <string>
@ -28,4 +28,4 @@ MediaType MediaTypeFromString(const std::string& type_str);
} // namespace cricket } // namespace cricket
#endif // WEBRTC_API_MEDIATYPES_H_ #endif // API_MEDIATYPES_H_

10
api/notifier.h
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@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_NOTIFIER_H_ #ifndef API_NOTIFIER_H_
#define WEBRTC_API_NOTIFIER_H_ #define API_NOTIFIER_H_
#include <list> #include <list>
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
namespace webrtc { namespace webrtc {
@ -58,4 +58,4 @@ class Notifier : public T {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_NOTIFIER_H_ #endif // API_NOTIFIER_H_

2
api/optional.cc
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@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree. * 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 rtc {
namespace optional_internal { namespace optional_internal {

12
api/optional.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_OPTIONAL_H_ #ifndef API_OPTIONAL_H_
#define WEBRTC_API_OPTIONAL_H_ #define API_OPTIONAL_H_
#include <algorithm> #include <algorithm>
#include <memory> #include <memory>
@ -20,9 +20,9 @@
#include <ostream> #include <ostream>
#endif // UNIT_TEST #endif // UNIT_TEST
#include "webrtc/api/array_view.h" #include "api/array_view.h"
#include "webrtc/rtc_base/checks.h" #include "rtc_base/checks.h"
#include "webrtc/rtc_base/sanitizer.h" #include "rtc_base/sanitizer.h"
namespace rtc { namespace rtc {
@ -404,4 +404,4 @@ void PrintTo(const rtc::Optional<T>& opt, std::ostream* os) {
} // namespace rtc } // namespace rtc
#endif // WEBRTC_API_OPTIONAL_H_ #endif // API_OPTIONAL_H_

4
api/optional_unittest.cc
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@ -14,8 +14,8 @@
#include <utility> #include <utility>
#include <vector> #include <vector>
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/rtc_base/gunit.h" #include "rtc_base/gunit.h"
namespace rtc { namespace rtc {

2
api/ortc/mediadescription.cc
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@ -8,6 +8,6 @@
* be found in the AUTHORS file in the root of the source tree. * 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 {} namespace webrtc {}

10
api/ortc/mediadescription.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_MEDIADESCRIPTION_H_ #ifndef API_ORTC_MEDIADESCRIPTION_H_
#define WEBRTC_API_ORTC_MEDIADESCRIPTION_H_ #define API_ORTC_MEDIADESCRIPTION_H_
#include <string> #include <string>
#include <utility> #include <utility>
#include <vector> #include <vector>
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/media/base/cryptoparams.h" #include "media/base/cryptoparams.h"
namespace webrtc { namespace webrtc {
@ -50,4 +50,4 @@ class MediaDescription {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_MEDIADESCRIPTION_H_ #endif // API_ORTC_MEDIADESCRIPTION_H_

4
api/ortc/mediadescription_unittest.cc
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/ortc/mediadescription.h" #include "api/ortc/mediadescription.h"
#include "webrtc/test/gtest.h" #include "test/gtest.h"
namespace webrtc { namespace webrtc {

38
api/ortc/ortcfactoryinterface.h
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@ -8,29 +8,29 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_ #ifndef API_ORTC_ORTCFACTORYINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_ #define API_ORTC_ORTCFACTORYINTERFACE_H_
#include <memory> #include <memory>
#include <string> #include <string>
#include <utility> // For std::move. #include <utility> // For std::move.
#include "webrtc/api/mediaconstraintsinterface.h" #include "api/mediaconstraintsinterface.h"
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h" #include "api/mediatypes.h"
#include "webrtc/api/ortc/ortcrtpreceiverinterface.h" #include "api/ortc/ortcrtpreceiverinterface.h"
#include "webrtc/api/ortc/ortcrtpsenderinterface.h" #include "api/ortc/ortcrtpsenderinterface.h"
#include "webrtc/api/ortc/packettransportinterface.h" #include "api/ortc/packettransportinterface.h"
#include "webrtc/api/ortc/rtptransportcontrollerinterface.h" #include "api/ortc/rtptransportcontrollerinterface.h"
#include "webrtc/api/ortc/rtptransportinterface.h" #include "api/ortc/rtptransportinterface.h"
#include "webrtc/api/ortc/srtptransportinterface.h" #include "api/ortc/srtptransportinterface.h"
#include "webrtc/api/ortc/udptransportinterface.h" #include "api/ortc/udptransportinterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/api/rtpparameters.h" #include "api/rtpparameters.h"
#include "webrtc/p2p/base/packetsocketfactory.h" #include "p2p/base/packetsocketfactory.h"
#include "webrtc/rtc_base/network.h" #include "rtc_base/network.h"
#include "webrtc/rtc_base/scoped_ref_ptr.h" #include "rtc_base/scoped_ref_ptr.h"
#include "webrtc/rtc_base/thread.h" #include "rtc_base/thread.h"
namespace webrtc { namespace webrtc {
@ -236,4 +236,4 @@ class OrtcFactoryInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCFACTORYINTERFACE_H_ #endif // API_ORTC_ORTCFACTORYINTERFACE_H_

16
api/ortc/ortcrtpreceiverinterface.h
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@ -15,14 +15,14 @@
// DtlsTransport. This is to allow different types of RTP transports (besides // DtlsTransport. This is to allow different types of RTP transports (besides
// DTLS-SRTP) to be used. // DTLS-SRTP) to be used.
#ifndef WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_ #ifndef API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_ #define API_ORTC_ORTCRTPRECEIVERINTERFACE_H_
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h" #include "api/mediatypes.h"
#include "webrtc/api/ortc/rtptransportinterface.h" #include "api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/api/rtpparameters.h" #include "api/rtpparameters.h"
namespace webrtc { namespace webrtc {
@ -81,4 +81,4 @@ class OrtcRtpReceiverInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCRTPRECEIVERINTERFACE_H_ #endif // API_ORTC_ORTCRTPRECEIVERINTERFACE_H_

16
api/ortc/ortcrtpsenderinterface.h
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@ -15,14 +15,14 @@
// DtlsTransport. This is to allow different types of RTP transports (besides // DtlsTransport. This is to allow different types of RTP transports (besides
// DTLS-SRTP) to be used. // DTLS-SRTP) to be used.
#ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_ #ifndef API_ORTC_ORTCRTPSENDERINTERFACE_H_
#define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_ #define API_ORTC_ORTCRTPSENDERINTERFACE_H_
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/mediatypes.h" #include "api/mediatypes.h"
#include "webrtc/api/ortc/rtptransportinterface.h" #include "api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/api/rtpparameters.h" #include "api/rtpparameters.h"
namespace webrtc { namespace webrtc {
@ -74,4 +74,4 @@ class OrtcRtpSenderInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_ #endif // API_ORTC_ORTCRTPSENDERINTERFACE_H_

6
api/ortc/packettransportinterface.h
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@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_ #ifndef API_ORTC_PACKETTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_ #define API_ORTC_PACKETTRANSPORTINTERFACE_H_
namespace rtc { namespace rtc {
@ -35,4 +35,4 @@ class PacketTransportInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_PACKETTRANSPORTINTERFACE_H_ #endif // API_ORTC_PACKETTRANSPORTINTERFACE_H_

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

14
api/ortc/rtptransportinterface.h
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@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ #ifndef API_ORTC_RTPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ #define API_ORTC_RTPTRANSPORTINTERFACE_H_
#include <string> #include <string>
#include "webrtc/api/optional.h" #include "api/optional.h"
#include "webrtc/api/ortc/packettransportinterface.h" #include "api/ortc/packettransportinterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/common_types.h" #include "common_types.h"
namespace webrtc { namespace webrtc {
@ -120,4 +120,4 @@ class RtpTransportInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_RTPTRANSPORTINTERFACE_H_ #endif // API_ORTC_RTPTRANSPORTINTERFACE_H_

2
api/ortc/sessiondescription.cc
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@ -8,6 +8,6 @@
* be found in the AUTHORS file in the root of the source tree. * 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 {} 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_SESSIONDESCRIPTION_H_ #ifndef API_ORTC_SESSIONDESCRIPTION_H_
#define WEBRTC_API_ORTC_SESSIONDESCRIPTION_H_ #define API_ORTC_SESSIONDESCRIPTION_H_
#include <string> #include <string>
#include <utility> #include <utility>
@ -42,4 +42,4 @@ class SessionDescription {
} // namespace webrtc } // 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include "webrtc/api/ortc/sessiondescription.h" #include "api/ortc/sessiondescription.h"
#include "webrtc/test/gtest.h" #include "test/gtest.h"
namespace webrtc { 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_ #ifndef API_ORTC_SRTPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_ #define API_ORTC_SRTPTRANSPORTINTERFACE_H_
#include "webrtc/api/ortc/rtptransportinterface.h" #include "api/ortc/rtptransportinterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/media/base/cryptoparams.h" #include "media/base/cryptoparams.h"
namespace webrtc { namespace webrtc {
@ -45,4 +45,4 @@ class SrtpTransportInterface : public RtpTransportInterface {
} // namespace webrtc } // 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_ #ifndef API_ORTC_UDPTRANSPORTINTERFACE_H_
#define WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_ #define API_ORTC_UDPTRANSPORTINTERFACE_H_
#include "webrtc/api/ortc/packettransportinterface.h" #include "api/ortc/packettransportinterface.h"
#include "webrtc/api/proxy.h" #include "api/proxy.h"
#include "webrtc/rtc_base/socketaddress.h" #include "rtc_base/socketaddress.h"
namespace webrtc { namespace webrtc {
@ -46,4 +46,4 @@ class UdpTransportInterface : public virtual PacketTransportInterface {
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_ORTC_UDPTRANSPORTINTERFACE_H_ #endif // API_ORTC_UDPTRANSPORTINTERFACE_H_

12
api/peerconnectionfactoryproxy.h
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@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ #ifndef API_PEERCONNECTIONFACTORYPROXY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ #define API_PEERCONNECTIONFACTORYPROXY_H_
#include <memory> #include <memory>
#include <string> #include <string>
#include <utility> #include <utility>
#include "webrtc/api/peerconnectioninterface.h" #include "api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h" #include "api/proxy.h"
#include "webrtc/rtc_base/bind.h" #include "rtc_base/bind.h"
namespace webrtc { namespace webrtc {
@ -75,4 +75,4 @@ END_PROXY_MAP()
} // namespace webrtc } // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_ #endif // API_PEERCONNECTIONFACTORYPROXY_H_

52
api/peerconnectioninterface.h
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@ -64,37 +64,37 @@
// 7. Once a candidate has been gathered, the PeerConnection will call the // 7. Once a candidate has been gathered, the PeerConnection will call the
// observer function OnIceCandidate. Send these candidates to the remote peer. // observer function OnIceCandidate. Send these candidates to the remote peer.
#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_ #ifndef API_PEERCONNECTIONINTERFACE_H_
#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_ #define API_PEERCONNECTIONINTERFACE_H_
#include <memory> #include <memory>
#include <string> #include <string>
#include <utility> #include <utility>
#include <vector> #include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h" #include "api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_encoder_factory.h"
#include "webrtc/api/datachannelinterface.h" #include "api/datachannelinterface.h"
#include "webrtc/api/dtmfsenderinterface.h" #include "api/dtmfsenderinterface.h"
#include "webrtc/api/jsep.h" #include "api/jsep.h"
#include "webrtc/api/mediastreaminterface.h" #include "api/mediastreaminterface.h"
#include "webrtc/api/rtcerror.h" #include "api/rtcerror.h"
#include "webrtc/api/rtpreceiverinterface.h" #include "api/rtpreceiverinterface.h"
#include "webrtc/api/rtpsenderinterface.h" #include "api/rtpsenderinterface.h"
#include "webrtc/api/stats/rtcstatscollectorcallback.h" #include "api/stats/rtcstatscollectorcallback.h"
#include "webrtc/api/statstypes.h" #include "api/statstypes.h"
#include "webrtc/api/umametrics.h" #include "api/umametrics.h"
#include "webrtc/call/callfactoryinterface.h" #include "call/callfactoryinterface.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h" #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "webrtc/media/base/mediachannel.h" #include "media/base/mediachannel.h"
#include "webrtc/media/base/videocapturer.h" #include "media/base/videocapturer.h"
#include "webrtc/p2p/base/portallocator.h" #include "p2p/base/portallocator.h"
#include "webrtc/rtc_base/fileutils.h" #include "rtc_base/fileutils.h"
#include "webrtc/rtc_base/network.h" #include "rtc_base/network.h"
#include "webrtc/rtc_base/rtccertificate.h" #include "rtc_base/rtccertificate.h"
#include "webrtc/rtc_base/rtccertificategenerator.h" #include "rtc_base/rtccertificategenerator.h"
#include "webrtc/rtc_base/socketaddress.h" #include "rtc_base/socketaddress.h"
#include "webrtc/rtc_base/sslstreamadapter.h" #include "rtc_base/sslstreamadapter.h"
namespace rtc { namespace rtc {
class SSLIdentity; class SSLIdentity;
@ -1224,4 +1224,4 @@ CreateModularPeerConnectionFactory(
} // namespace webrtc } // 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. * be found in the AUTHORS file in the root of the source tree.
*/ */
#ifndef WEBRTC_API_PEERCONNECTIONPROXY_H_ #ifndef API_PEERCONNECTIONPROXY_H_
#define WEBRTC_API_PEERCONNECTIONPROXY_H_ #define API_PEERCONNECTIONPROXY_H_
#include <string> #include <string>
#include <vector> #include <vector>
#include "webrtc/api/peerconnectioninterface.h" #include "api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h" #include "api/proxy.h"
namespace webrtc { namespace webrtc {
@ -111,4 +111,4 @@ END_PROXY_MAP()
} // namespace webrtc } // 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 // The variant defined with BEGIN_OWNED_PROXY_MAP does not use
// refcounting, and instead just takes ownership of the object being proxied. // refcounting, and instead just takes ownership of the object being proxied.
#ifndef WEBRTC_API_PROXY_H_ #ifndef API_PROXY_H_
#define WEBRTC_API_PROXY_H_ #define API_PROXY_H_
#include <memory> #include <memory>
#include <utility> #include <utility>
#include "webrtc/rtc_base/event.h" #include "rtc_base/event.h"
#include "webrtc/rtc_base/thread.h" #include "rtc_base/thread.h"
namespace webrtc { namespace webrtc {
@ -579,4 +579,4 @@ class MethodCall5 : public rtc::Message,
} // namespace webrtc } // 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. * 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 { namespace {

10
api/rtcerror.h
View File

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

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