mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
Merge remote-tracking branch 'upstream/branch-heads/6834' into mutexlox/m132-merge
This commit is contained in:
Miriam Zimmerman
commit
8604cca7c6
1199 changed files with 17877 additions and 27545 deletions
4
.gn
4
.gn
|
|
@ -83,4 +83,8 @@ default_args = {
|
|||
# WebRTC doesn't use jni_zero's multiplexing. Since this causes an error
|
||||
# let's temporarily disable it.
|
||||
enable_jni_multiplexing = false
|
||||
|
||||
# TODO(b/42223878): use_fuzztest_wrapper adds a dependency to //base so
|
||||
# let's temporarly disable it.
|
||||
use_fuzztest_wrapper = false
|
||||
}
|
||||
|
|
|
|||
4
AUTHORS
4
AUTHORS
|
|
@ -53,6 +53,7 @@ Eike Rathke <erathke@redhat.com>
|
|||
Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
|
||||
Filip Hlasek <filip@orcamobility.ai>
|
||||
Frederik Riedel, Frogg GmbH <frederik.riedel@frogg.io>
|
||||
Gao Chun <gaochun.dev@gmail.com>
|
||||
Giji Gangadharan <giji.g@samsung.com>
|
||||
Graham Yoakum <gyoakum@skobalt.com>
|
||||
Gustavo Garcia <gustavogb@gmail.com>
|
||||
|
|
@ -122,7 +123,9 @@ Satender Saroha <ssaroha@yahoo.com>
|
|||
Saul Kravitz <Saul.Kravitz@celera.com>
|
||||
Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
|
||||
Shaofan Qi <vshaqi@gmail.com>
|
||||
Shigemasa Watanabe <shigemasa7watanabe@gmail.com>
|
||||
Shuhai Peng <shuhai.peng@intel.com>
|
||||
Shunbo Li <lishunbo@shengwang.cn>
|
||||
Seija <doremylover123@gmail.com>
|
||||
Silviu Caragea <silviu.cpp@gmail.com>
|
||||
Stefan Gula <steweg@gmail.com>
|
||||
|
|
@ -149,6 +152,7 @@ Yuriy Pavlyshak <yuriy@appear.in>
|
|||
Yusuke Suzuki <utatane.tea@gmail.com>
|
||||
Pengfei Han <hanpfei@gmail.com>
|
||||
Yingying Ma <yingying.ma@intel.com>
|
||||
Hailin Zhao <lynnworld7@gmail.com>
|
||||
# END individuals section.
|
||||
|
||||
# BEGIN organizations section.
|
||||
|
|
|
|||
22
BUILD.gn
22
BUILD.gn
|
|
@ -419,19 +419,11 @@ config("common_config") {
|
|||
}
|
||||
|
||||
if (is_clang) {
|
||||
cflags += [ "-Wc++11-narrowing" ]
|
||||
|
||||
if (!is_fuchsia) {
|
||||
# Compiling with the Fuchsia SDK results in Wundef errors
|
||||
# TODO(bugs.fuchsia.dev/100722): Remove from (!is_fuchsia) branch when
|
||||
# Fuchsia build errors are fixed.
|
||||
cflags += [ "-Wundef" ]
|
||||
}
|
||||
|
||||
if (!is_nacl) {
|
||||
# Flags NaCl (Clang 3.7) do not recognize.
|
||||
cflags += [ "-Wunused-lambda-capture" ]
|
||||
}
|
||||
cflags += [
|
||||
"-Wc++11-narrowing",
|
||||
"-Wundef",
|
||||
"-Wunused-lambda-capture",
|
||||
]
|
||||
}
|
||||
|
||||
if (is_win && !is_clang) {
|
||||
|
|
@ -656,7 +648,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
deps = [
|
||||
"api:compile_all_headers",
|
||||
"api:rtc_api_unittests",
|
||||
"api/audio/test:audio_api_unittests",
|
||||
"api/audio:audio_api_unittests",
|
||||
"api/audio_codecs/test:audio_codecs_api_unittests",
|
||||
"api/numerics:numerics_unittests",
|
||||
"api/task_queue:pending_task_safety_flag_unittests",
|
||||
|
|
@ -672,6 +664,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
"p2p:rtc_p2p_unittests",
|
||||
"rtc_base:async_dns_resolver_unittests",
|
||||
"rtc_base:async_packet_socket_unittest",
|
||||
"rtc_base:async_udp_socket_unittest",
|
||||
"rtc_base:callback_list_unittests",
|
||||
"rtc_base:rtc_base_approved_unittests",
|
||||
"rtc_base:rtc_base_unittests",
|
||||
|
|
@ -731,6 +724,7 @@ if (rtc_include_tests && !build_with_chromium) {
|
|||
|
||||
# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
|
||||
video_engine_tests_resources = [
|
||||
"resources/ConferenceMotion_1280_720_50.yuv",
|
||||
"resources/foreman_cif_short.yuv",
|
||||
"resources/voice_engine/audio_long16.pcm",
|
||||
]
|
||||
|
|
|
|||
151
DEPS
151
DEPS
|
|
@ -10,7 +10,7 @@ vars = {
|
|||
# chromium waterfalls. More info at: crbug.com/570091.
|
||||
'checkout_configuration': 'default',
|
||||
'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
|
||||
'chromium_revision': '3b70d6f26cf7a1f53bcf227a621506727653902f',
|
||||
'chromium_revision': '030af8fbf08f2bb06397304628f15b2d3bdb90d0',
|
||||
|
||||
# Fetch the prebuilt binaries for llvm-cov and llvm-profdata. Needed to
|
||||
# process the raw profiles produced by instrumented targets (built with
|
||||
|
|
@ -20,6 +20,9 @@ vars = {
|
|||
# Fetch clangd into the same bin/ directory as our clang binary.
|
||||
'checkout_clangd': False,
|
||||
|
||||
# Fetch libraries required to compile and run fuzzer tests.
|
||||
'checkout_fuzzer': False,
|
||||
|
||||
'chromium_git': 'https://chromium.googlesource.com',
|
||||
|
||||
# Keep the Chromium default of generating location tags.
|
||||
|
|
@ -30,7 +33,7 @@ vars = {
|
|||
|
||||
# By default, download the fuchsia sdk from the public sdk directory.
|
||||
'fuchsia_sdk_cipd_prefix': 'fuchsia/sdk/core/',
|
||||
'fuchsia_version': 'version:24.20240906.3.1',
|
||||
'fuchsia_version': 'version:24.20241014.3.1',
|
||||
# By default, download the fuchsia images from the fuchsia GCS bucket.
|
||||
'fuchsia_images_bucket': 'fuchsia',
|
||||
'checkout_fuchsia': False,
|
||||
|
|
@ -45,14 +48,14 @@ vars = {
|
|||
# RBE instance to use for running remote builds
|
||||
'rbe_instance': 'projects/rbe-webrtc-developer/instances/default_instance',
|
||||
# reclient CIPD package version
|
||||
'reclient_version': 're_client_version:0.163.0.d27158ab-gomaip',
|
||||
'reclient_version': 're_client_version:0.168.0.c46e68bc-gomaip',
|
||||
|
||||
# ninja CIPD package.
|
||||
'ninja_package': 'infra/3pp/tools/ninja/',
|
||||
|
||||
# ninja CIPD package version
|
||||
# https://chrome-infra-packages.appspot.com/p/infra/3pp/tools/ninja
|
||||
'ninja_version': 'version:2@1.12.1.chromium.4',
|
||||
'ninja_version': 'version:3@1.12.1.chromium.4',
|
||||
|
||||
# condition to allowlist deps for non-git-source processing.
|
||||
'non_git_source': 'True',
|
||||
|
|
@ -71,30 +74,30 @@ deps = {
|
|||
|
||||
# TODO(kjellander): Move this to be Android-only.
|
||||
'src/base':
|
||||
'https://chromium.googlesource.com/chromium/src/base@7f2a56694b330c2cb2954eee78469ffc69952448',
|
||||
'https://chromium.googlesource.com/chromium/src/base@69f3676cdbd05660c136a6cc3b23383cbabe53d8',
|
||||
'src/build':
|
||||
'https://chromium.googlesource.com/chromium/src/build@8e0612cd2c4e4d7c15a456384760815bdbd64ef1',
|
||||
'https://chromium.googlesource.com/chromium/src/build@05874e6c9429039f1747034af61de1fe2ea1ed06',
|
||||
'src/buildtools':
|
||||
'https://chromium.googlesource.com/chromium/src/buildtools@e4f992c0660e6cee637c7b97aabf04d0a58d5555',
|
||||
'https://chromium.googlesource.com/chromium/src/buildtools@db0eae9640184fb132061f248f6108771a6ea2d4',
|
||||
# Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.
|
||||
'src/examples/androidtests/third_party/gradle': {
|
||||
'url': 'https://chromium.googlesource.com/external/github.com/gradle/gradle.git@f2d1fb54a951d8b11d25748e4711bec8d128d7e3',
|
||||
'condition': 'checkout_android',
|
||||
},
|
||||
'src/ios': {
|
||||
'url': 'https://chromium.googlesource.com/chromium/src/ios@8e01efd2a9dba90f0094ed0fa908a6e907410d74',
|
||||
'url': 'https://chromium.googlesource.com/chromium/src/ios@452c8ab2166e50fe43484dd3da5e19ce14a68e6b',
|
||||
'condition': 'checkout_ios',
|
||||
},
|
||||
'src/testing':
|
||||
'https://chromium.googlesource.com/chromium/src/testing@3f4138fcb15362469527d9310e21600dfdfab691',
|
||||
'https://chromium.googlesource.com/chromium/src/testing@eac4c18f749488d7232af275405e0e1b67d84cb3',
|
||||
'src/third_party':
|
||||
'https://chromium.googlesource.com/chromium/src/third_party@27c9b4efc25d7ae5479b884a7d818058c6066d92',
|
||||
'https://chromium.googlesource.com/chromium/src/third_party@9d80a193fbfe8e2f2a29be2efeee93de9a826dcb',
|
||||
|
||||
'src/buildtools/linux64': {
|
||||
'packages': [
|
||||
{
|
||||
'package': 'gn/gn/linux-${{arch}}',
|
||||
'version': 'git_revision:20806f79c6b4ba295274e3a589d85db41a02fdaa',
|
||||
'version': 'git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a',
|
||||
}
|
||||
],
|
||||
'dep_type': 'cipd',
|
||||
|
|
@ -104,7 +107,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'gn/gn/mac-${{arch}}',
|
||||
'version': 'git_revision:20806f79c6b4ba295274e3a589d85db41a02fdaa',
|
||||
'version': 'git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a',
|
||||
}
|
||||
],
|
||||
'dep_type': 'cipd',
|
||||
|
|
@ -114,7 +117,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'gn/gn/windows-amd64',
|
||||
'version': 'git_revision:20806f79c6b4ba295274e3a589d85db41a02fdaa',
|
||||
'version': 'git_revision:feafd1012a32c05ec6095f69ddc3850afb621f3a',
|
||||
}
|
||||
],
|
||||
'dep_type': 'cipd',
|
||||
|
|
@ -133,14 +136,56 @@ deps = {
|
|||
'condition': 'not (host_os == "linux" and host_cpu == "arm64")',
|
||||
},
|
||||
|
||||
'third_party/rust': {
|
||||
'url': Var('chromium_git') + '/chromium/src/third_party/rust@0e0ef14876a32128574eaf80bc7fc1c7cde92006',
|
||||
'condition': 'checkout_android or checkout_fuzzer',
|
||||
},
|
||||
|
||||
'third_party/rust-toolchain': {
|
||||
'dep_type': 'gcs',
|
||||
'bucket': 'chromium-browser-clang',
|
||||
'objects': [
|
||||
{
|
||||
'object_name': 'Linux_x64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz',
|
||||
'sha256sum': '043bc520520424fad9fdfc87102e3af7c8f1ca4da3f7885e18f54fd29fc783e4',
|
||||
'size_bytes': 115068636,
|
||||
'generation': 1725552307196828,
|
||||
'condition': 'host_os == "linux" and non_git_source',
|
||||
},
|
||||
{
|
||||
'object_name': 'Mac/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz',
|
||||
'sha256sum': '100befb02da0940fdede41a0ad0e28ded1ce4b6794b5e488bccc91ba1d27348a',
|
||||
'size_bytes': 108415524,
|
||||
'generation': 1725552308530261,
|
||||
'condition': 'host_os == "mac" and host_cpu == "x64"',
|
||||
},
|
||||
{
|
||||
'object_name': 'Mac_arm64/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz',
|
||||
'sha256sum': '6f3ba8fe5a9590b476c087b412bd60e7e46ec59bc37aac76d9f7894c6bc34a75',
|
||||
'size_bytes': 97820200,
|
||||
'generation': 1725552309875662,
|
||||
'condition': 'host_os == "mac" and host_cpu == "arm64"',
|
||||
},
|
||||
{
|
||||
'object_name': 'Win/rust-toolchain-009e73825af0e59ad4fc603562e038b3dbd6593a-2-llvmorg-20-init-3847-g69c43468.tar.xz',
|
||||
'sha256sum': '1d04274c7a96aa0148a9bb5b29fa5b33c24b97971e33634bed62ce504b1cb346',
|
||||
'size_bytes': 172453420,
|
||||
'generation': 1725552311180142,
|
||||
'condition': 'host_os == "win"',
|
||||
},
|
||||
],
|
||||
},
|
||||
|
||||
'src/third_party/clang-format/script':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/clang/tools/clang-format.git@3c0acd2d4e73dd911309d9e970ba09d58bf23a62',
|
||||
'src/third_party/libc++/src':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@887f732df07adac7a67997ff0a91baa12081e5a2',
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxx.git@6a68fd412b9aecd515a20a7cf84d11b598bfaf96',
|
||||
'src/third_party/libc++abi/src':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@29b2e9a0f48688da116692cb04758393053d269c',
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libcxxabi.git@9a1d90c3b412d5ebeb97a6e33d98e1d0dd923221',
|
||||
'src/third_party/llvm-libc/src':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libc.git@98ed09318387deb45282ca1c92a384499860b76a',
|
||||
'src/third_party/libunwind/src':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@dc70138c3e68e2f946585f134e20815851e26263',
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/libunwind.git@efc3baa2d1ece3630fcfa72bef93ed831bcaec4c',
|
||||
|
||||
'src/third_party/test_fonts/test_fonts': {
|
||||
'dep_type': 'gcs',
|
||||
|
|
@ -214,7 +259,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/android_build_tools/error_prone',
|
||||
'version': 'fNCLAzE8NSvOXTryvUGT3NmX8no8lyRHR1yfY0zbv8YC',
|
||||
'version': 'hUxlP8GvC1xhmZ6r9xjYau2laPlzHbs_P2emx4ZL4jgC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -244,11 +289,11 @@ deps = {
|
|||
},
|
||||
|
||||
'src/third_party/boringssl/src':
|
||||
'https://boringssl.googlesource.com/boringssl.git@e724ef02089bf2bb494203231fc5cb62acc2fad6',
|
||||
'https://boringssl.googlesource.com/boringssl.git@0fa9014d861f024e28fb491ed7b1cfd44103dee7',
|
||||
'src/third_party/breakpad/breakpad':
|
||||
'https://chromium.googlesource.com/breakpad/breakpad.git@6b0c5b7ee1988a14a4af94564e8ae8bba8a94374',
|
||||
'src/third_party/catapult':
|
||||
'https://chromium.googlesource.com/catapult.git@296226a4a0067c8cffeb8831fb87526a8035f3cc',
|
||||
'https://chromium.googlesource.com/catapult.git@44791916611acec1cd74c492c7453e46d9b0dbd2',
|
||||
'src/third_party/ced/src': {
|
||||
'url': 'https://chromium.googlesource.com/external/github.com/google/compact_enc_det.git@ba412eaaacd3186085babcd901679a48863c7dd5',
|
||||
},
|
||||
|
|
@ -261,9 +306,9 @@ deps = {
|
|||
'src/third_party/crc32c/src':
|
||||
'https://chromium.googlesource.com/external/github.com/google/crc32c.git@d3d60ac6e0f16780bcfcc825385e1d338801a558',
|
||||
'src/third_party/depot_tools':
|
||||
'https://chromium.googlesource.com/chromium/tools/depot_tools.git@d43a3eb748e40838c41e7a3ca53a3c6cbee0ea7a',
|
||||
'https://chromium.googlesource.com/chromium/tools/depot_tools.git@20b9bdcace7ed561d6a75728c85373503473cb6b',
|
||||
'src/third_party/ffmpeg':
|
||||
'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@91903c28af60a732a051c343b496e1188eec9b05',
|
||||
'https://chromium.googlesource.com/chromium/third_party/ffmpeg.git@686d6944501a6ee9c849581e3fe343273d4af3f6',
|
||||
'src/third_party/flatbuffers/src':
|
||||
'https://chromium.googlesource.com/external/github.com/google/flatbuffers.git@8db59321d9f02cdffa30126654059c7d02f70c32',
|
||||
'src/third_party/grpc/src': {
|
||||
|
|
@ -275,7 +320,7 @@ deps = {
|
|||
'condition': 'checkout_linux',
|
||||
},
|
||||
'src/third_party/freetype/src':
|
||||
'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@83af801b552111e37d9466a887e1783a0fb5f196',
|
||||
'https://chromium.googlesource.com/chromium/src/third_party/freetype2.git@f02bffad0fd57f3acfa835c3f2899c5b71ff8be0',
|
||||
'src/third_party/harfbuzz-ng/src':
|
||||
'https://chromium.googlesource.com/external/github.com/harfbuzz/harfbuzz.git@1da053e87f0487382404656edca98b85fe51f2fd',
|
||||
'src/third_party/google_benchmark/src': {
|
||||
|
|
@ -289,9 +334,9 @@ deps = {
|
|||
'condition': 'checkout_android',
|
||||
},
|
||||
'src/third_party/googletest/src':
|
||||
'https://chromium.googlesource.com/external/github.com/google/googletest.git@0953a17a4281fc26831da647ad3fcd5e21e6473b',
|
||||
'https://chromium.googlesource.com/external/github.com/google/googletest.git@62df7bdbc10887e094661e07ec2595b7920376fd',
|
||||
'src/third_party/icu': {
|
||||
'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@9408c6fd4a39e6fef0e1c4077602e1c83b15f3fb',
|
||||
'url': 'https://chromium.googlesource.com/chromium/deps/icu.git@4239b1559d11d4fa66c100543eda4161e060311e',
|
||||
},
|
||||
'src/third_party/jdk/current': {
|
||||
'packages': [
|
||||
|
|
@ -326,7 +371,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/kotlin_stdlib',
|
||||
'version': 'b48PGccyaRl9kI3MLdJ36I-z5Zcj9NOCjsUg02Bl-EIC',
|
||||
'version': 'XJ7_doI-Qt7GFaSQ9BNo-3qF7Gv2--9Sa8GEUdjxMTUC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -337,37 +382,39 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/kotlinc',
|
||||
'version': 'PKwXlOC2iNJx0ELcmUpSTk0GHbNMEM-HZwhs2AWl1WUC',
|
||||
'version': 'FNZSCjJ6yKsi6oRcgQrt-lX0MDlaWoxT7gPTz0CjLhMC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
'dep_type': 'cipd',
|
||||
},
|
||||
# Used for building libFuzzers (only supports Linux).
|
||||
'src/third_party/libFuzzer/src':
|
||||
'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@487e79376394754705984c5de7c4ce7f82f2bd7c',
|
||||
'src/third_party/libFuzzer/src': {
|
||||
'url': 'https://chromium.googlesource.com/external/github.com/llvm/llvm-project/compiler-rt/lib/fuzzer.git@487e79376394754705984c5de7c4ce7f82f2bd7c',
|
||||
'condition': 'checkout_fuzzer',
|
||||
},
|
||||
'src/third_party/fuzztest/src':
|
||||
'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@a29e31cb00ec9b123dec5a0c6b8d4bc12c2480c8',
|
||||
'https://chromium.googlesource.com/external/github.com/google/fuzztest.git@0021f30508bc7f73fa5270962d022acb480d242f',
|
||||
'src/third_party/libjpeg_turbo':
|
||||
'https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git@927aabfcd26897abb9776ecf2a6c38ea5bb52ab6',
|
||||
'src/third_party/libsrtp':
|
||||
'https://chromium.googlesource.com/chromium/deps/libsrtp.git@000edd791434c8738455f10e0dd6b268a4852c0b',
|
||||
'src/third_party/dav1d/libdav1d':
|
||||
'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@6b3c489a2ee2c030f351f21987c27611b4cbe725',
|
||||
'https://chromium.googlesource.com/external/github.com/videolan/dav1d.git@389450f61ea0b2057fc9ea393d3065859c4ba7f2',
|
||||
'src/third_party/libaom/source/libaom':
|
||||
'https://aomedia.googlesource.com/aom.git@d5265b173616ce62de231cd1b1eae853ad03641e',
|
||||
'https://aomedia.googlesource.com/aom.git@840f8797871cc587f7113ea9d2483a1156d57c0e',
|
||||
'src/third_party/libunwindstack': {
|
||||
'url': 'https://chromium.googlesource.com/chromium/src/third_party/libunwindstack.git@a3bb4cd02e0e984a235069f812cbef2b37c389e5',
|
||||
'condition': 'checkout_android',
|
||||
},
|
||||
'src/third_party/perfetto':
|
||||
'https://android.googlesource.com/platform/external/perfetto.git@97a5d7ed4d7ddb3844a5cbd184e241b99fe6ebcc',
|
||||
'https://android.googlesource.com/platform/external/perfetto.git@e57316a6ae4e85e9bc8ba82a6c6a93eb5d9a72aa',
|
||||
'src/third_party/protobuf-javascript/src':
|
||||
Var('chromium_git') + '/external/github.com/protocolbuffers/protobuf-javascript' + '@' + 'e34549db516f8712f678fcd4bc411613b5cc5295',
|
||||
'src/third_party/libvpx/source/libvpx':
|
||||
'https://chromium.googlesource.com/webm/libvpx.git@fbf63dff1f528d44f24bd662abb89fd01a4a1c25',
|
||||
'https://chromium.googlesource.com/webm/libvpx.git@906334ac1de2b0afa666472dce5545b82c1251fb',
|
||||
'src/third_party/libyuv':
|
||||
'https://chromium.googlesource.com/libyuv/libyuv.git@874f391dbf13dcc84e13a494daed57539ffe2935',
|
||||
'https://chromium.googlesource.com/libyuv/libyuv.git@a8e59d207483f75b87dd5fc670e937672cdf5776',
|
||||
'src/third_party/lss': {
|
||||
'url': 'https://chromium.googlesource.com/linux-syscall-support.git@ce877209e11aa69dcfffbd53ef90ea1d07136521',
|
||||
'condition': 'checkout_android or checkout_linux',
|
||||
|
|
@ -396,7 +443,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/r8',
|
||||
'version': 'mqzE1g-dbf23PQ04w-4TpmupGkMOnZljVa993-MkzhIC',
|
||||
'version': '-i5fwP_NzM6Ylg5AsSGEotYN7hQgV852gXCslvXIrRwC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -409,7 +456,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/r8',
|
||||
'version': 'vw5kLlW3-suSlCKSO9OQpFWpR8oDnvQ8k1RgKNUapQYC',
|
||||
'version': '3KCj5eRYCvGGYs5i90pRaeihkzsqgUGc4OkICT8AOlIC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -420,7 +467,7 @@ deps = {
|
|||
'condition': 'checkout_android',
|
||||
},
|
||||
'src/tools':
|
||||
'https://chromium.googlesource.com/chromium/src/tools@453f7ad001cd0d341ecb437eb164fd6c4b5cc08d',
|
||||
'https://chromium.googlesource.com/chromium/src/tools@c809c4133185094fbd75508202221ff14fc92f7c',
|
||||
|
||||
'src/third_party/espresso': {
|
||||
'packages': [
|
||||
|
|
@ -459,7 +506,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/androidx',
|
||||
'version': '-HPg6kSZtheePVbklOqBtMGHIYS9d0uoR7S-NfR5kGoC',
|
||||
'version': 'k1wif7sS51pJGSFGN7FAeGWDorxgPart9E1f383TQL4C',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android and non_git_source',
|
||||
|
|
@ -470,7 +517,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/android_build_tools/manifest_merger',
|
||||
'version': 'JnUHT3SsYpv4WBlOV5VrBkuj2Q9SJ3cXlNHZ-u78L5IC',
|
||||
'version': 'rnIeJMlGw7adxOKZofLsm7tdYaOy1nHivJn9ck7ocVkC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -541,7 +588,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/turbine',
|
||||
'version': 'aLKJszAm9PXh5T9KNcDH0MQJeEAIpEiFAYVEimlKdDsC',
|
||||
'version': 'vSia3h9tzpwpP_goLj4HMdl7_FEB5iVCv9nU5ZXOfIMC',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android',
|
||||
|
|
@ -552,11 +599,11 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'infra/tools/luci/isolate/${{platform}}',
|
||||
'version': 'git_revision:1aca70b6bf116c1bd8fbf0526c9a89e9be308718',
|
||||
'version': 'git_revision:7dd39503276dfa4a920102ca77a2f409f2f67655',
|
||||
},
|
||||
{
|
||||
'package': 'infra/tools/luci/swarming/${{platform}}',
|
||||
'version': 'git_revision:1aca70b6bf116c1bd8fbf0526c9a89e9be308718',
|
||||
'version': 'git_revision:7dd39503276dfa4a920102ca77a2f409f2f67655',
|
||||
},
|
||||
],
|
||||
'dep_type': 'cipd',
|
||||
|
|
@ -881,7 +928,7 @@ deps = {
|
|||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/android_deps/libs/com_google_android_material_material',
|
||||
'version': 'version:2@1.11.0-beta01.cr1',
|
||||
'version': 'version:2@1.13.0-alpha05.cr1',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android and non_git_source',
|
||||
|
|
@ -1559,6 +1606,17 @@ deps = {
|
|||
'dep_type': 'cipd',
|
||||
},
|
||||
|
||||
'src/third_party/android_deps/cipd/libs/org_jspecify_jspecify': {
|
||||
'packages': [
|
||||
{
|
||||
'package': 'chromium/third_party/android_deps/libs/org_jspecify_jspecify',
|
||||
'version': 'version:2@1.0.0.cr1',
|
||||
},
|
||||
],
|
||||
'condition': 'checkout_android and non_git_source',
|
||||
'dep_type': 'cipd',
|
||||
},
|
||||
|
||||
'src/third_party/android_deps/cipd/libs/org_mockito_mockito_android': {
|
||||
'packages': [
|
||||
{
|
||||
|
|
@ -1952,6 +2010,12 @@ hooks = [
|
|||
'action': ['vpython3', 'src/tools/clang/scripts/update.py',
|
||||
'--package=clangd'],
|
||||
},
|
||||
{
|
||||
'name': 'rust',
|
||||
'pattern': '.',
|
||||
'condition': 'checkout_android or checkout_fuzzer',
|
||||
'action': ['python3', 'src/tools/rust/update_rust.py'],
|
||||
},
|
||||
{
|
||||
# Update LASTCHANGE.
|
||||
'name': 'lastchange',
|
||||
|
|
@ -2124,6 +2188,7 @@ include_rules = [
|
|||
"+absl/strings/ascii.h",
|
||||
"+absl/strings/escaping.h",
|
||||
"+absl/strings/match.h",
|
||||
"+absl/strings/str_cat.h", # note - allowed for single argument version only
|
||||
"+absl/strings/str_replace.h",
|
||||
"+absl/strings/string_view.h",
|
||||
"+absl/types/variant.h",
|
||||
|
|
|
|||
80
PRESUBMIT.py
80
PRESUBMIT.py
|
|
@ -168,7 +168,8 @@ API_CHANGE_MSG = """
|
|||
You seem to be changing native API header files. Please make sure that you:
|
||||
1. Make compatible changes that don't break existing clients. Usually
|
||||
this is done by keeping the existing method signatures unchanged.
|
||||
2. Mark the old stuff as deprecated (use the ABSL_DEPRECATED macro).
|
||||
2. Mark the old stuff as deprecated (use the [[deprecated]] attribute or
|
||||
the ABSL_DEPRECATE_AND_INLINE macro).
|
||||
3. Create a timeline and plan for when the deprecated stuff will be
|
||||
removed. (The amount of time we give users to change their code
|
||||
should be informed by how much work it is for them. If they just
|
||||
|
|
@ -186,7 +187,7 @@ Related files:
|
|||
|
||||
def CheckNativeApiHeaderChanges(input_api, output_api):
|
||||
"""Checks to remind proper changing of native APIs."""
|
||||
files = []
|
||||
files = set()
|
||||
source_file_filter = lambda x: input_api.FilterSourceFile(
|
||||
x, files_to_check=[r'.+\.(gn|gni|h)$'])
|
||||
for f in input_api.AffectedSourceFiles(source_file_filter):
|
||||
|
|
@ -195,20 +196,20 @@ def CheckNativeApiHeaderChanges(input_api, output_api):
|
|||
if path == 'api':
|
||||
# Special case: Subdirectories included.
|
||||
if dn == 'api' or dn.startswith('api/'):
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
else:
|
||||
# Normal case: Subdirectories not included.
|
||||
if dn == path:
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
|
||||
if files:
|
||||
return [output_api.PresubmitNotifyResult(API_CHANGE_MSG, files)]
|
||||
return [output_api.PresubmitNotifyResult(API_CHANGE_MSG, list(files))]
|
||||
return []
|
||||
|
||||
|
||||
def CheckNoIOStreamInHeaders(input_api, output_api, source_file_filter):
|
||||
"""Checks to make sure no .h files include <iostream>."""
|
||||
files = []
|
||||
files = set()
|
||||
pattern = input_api.re.compile(r'^#include\s*<iostream>',
|
||||
input_api.re.MULTILINE)
|
||||
file_filter = lambda x: (input_api.FilterSourceFile(x) and
|
||||
|
|
@ -218,7 +219,7 @@ def CheckNoIOStreamInHeaders(input_api, output_api, source_file_filter):
|
|||
continue
|
||||
contents = input_api.ReadFile(f)
|
||||
if pattern.search(contents):
|
||||
files.append(f)
|
||||
files.add(f)
|
||||
|
||||
if len(files) > 0:
|
||||
return [
|
||||
|
|
@ -226,14 +227,14 @@ def CheckNoIOStreamInHeaders(input_api, output_api, source_file_filter):
|
|||
'Do not #include <iostream> in header files, since it inserts '
|
||||
'static initialization into every file including the header. '
|
||||
'Instead, #include <ostream>. See http://crbug.com/94794',
|
||||
files)
|
||||
list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
||||
def CheckNoPragmaOnce(input_api, output_api, source_file_filter):
|
||||
"""Make sure that banned functions are not used."""
|
||||
files = []
|
||||
files = set()
|
||||
pattern = input_api.re.compile(r'^#pragma\s+once', input_api.re.MULTILINE)
|
||||
file_filter = lambda x: (input_api.FilterSourceFile(x) and
|
||||
source_file_filter(x))
|
||||
|
|
@ -242,14 +243,14 @@ def CheckNoPragmaOnce(input_api, output_api, source_file_filter):
|
|||
continue
|
||||
contents = input_api.ReadFile(f)
|
||||
if pattern.search(contents):
|
||||
files.append(f)
|
||||
files.add(f)
|
||||
|
||||
if files:
|
||||
return [
|
||||
output_api.PresubmitError(
|
||||
'Do not use #pragma once in header files.\n'
|
||||
'See http://www.chromium.org/developers/coding-style'
|
||||
'#TOC-File-headers', files)
|
||||
'#TOC-File-headers', list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -316,12 +317,12 @@ def CheckApprovedFilesLintClean(input_api, output_api,
|
|||
# work with not-yet-converted code, we're only applying it to new (or
|
||||
# moved/renamed) files and files not listed in CPPLINT_EXCEPTIONS.
|
||||
verbosity_level = 1
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedSourceFiles(source_file_filter):
|
||||
# Note that moved/renamed files also count as added.
|
||||
if f.Action() == 'A' or not IsLintDisabled(disabled_paths,
|
||||
f.LocalPath()):
|
||||
files.append(f.AbsoluteLocalPath())
|
||||
files.add(f.AbsoluteLocalPath())
|
||||
|
||||
for file_name in files:
|
||||
cpplint.ProcessFile(file_name, verbosity_level)
|
||||
|
|
@ -342,7 +343,7 @@ def CheckNoSourcesAbove(input_api, gn_files, output_api):
|
|||
re.MULTILINE | re.DOTALL)
|
||||
file_pattern = input_api.re.compile(r'"((\.\./.*?)|(//.*?))"')
|
||||
violating_gn_files = set()
|
||||
violating_source_entries = []
|
||||
violating_source_entries = set()
|
||||
for gn_file in gn_files:
|
||||
contents = input_api.ReadFile(gn_file)
|
||||
for source_block_match in source_pattern.finditer(contents):
|
||||
|
|
@ -352,7 +353,7 @@ def CheckNoSourcesAbove(input_api, gn_files, output_api):
|
|||
source_block_match.group(1)):
|
||||
source_file = file_list_match.group(1)
|
||||
if 'overrides/' not in source_file:
|
||||
violating_source_entries.append(source_file)
|
||||
violating_source_entries.add(source_file)
|
||||
violating_gn_files.add(gn_file)
|
||||
if violating_gn_files:
|
||||
return [
|
||||
|
|
@ -363,7 +364,7 @@ def CheckNoSourcesAbove(input_api, gn_files, output_api):
|
|||
'Invalid source entries:\n'
|
||||
'%s\n'
|
||||
'Violating GN files:' % '\n'.join(violating_source_entries),
|
||||
items=violating_gn_files)
|
||||
items=list(violating_gn_files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -622,9 +623,9 @@ def CheckGnChanges(input_api, output_api):
|
|||
files_to_check=(r'.+\.(gn|gni)$', ),
|
||||
files_to_skip=(r'.*/presubmit_checks_lib/testdata/.*', )))
|
||||
|
||||
gn_files = []
|
||||
gn_files = set()
|
||||
for f in input_api.AffectedSourceFiles(file_filter):
|
||||
gn_files.append(f)
|
||||
gn_files.add(f)
|
||||
|
||||
result = []
|
||||
if gn_files:
|
||||
|
|
@ -840,7 +841,7 @@ def RunPythonTests(input_api, output_api):
|
|||
def CheckUsageOfGoogleProtobufNamespace(input_api, output_api,
|
||||
source_file_filter):
|
||||
"""Checks that the namespace google::protobuf has not been used."""
|
||||
files = []
|
||||
files = set()
|
||||
pattern = input_api.re.compile(r'google::protobuf')
|
||||
proto_utils_path = os.path.join('rtc_base', 'protobuf_utils.h')
|
||||
file_filter = lambda x: (input_api.FilterSourceFile(x) and
|
||||
|
|
@ -850,14 +851,14 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api,
|
|||
continue
|
||||
contents = input_api.ReadFile(f)
|
||||
if pattern.search(contents):
|
||||
files.append(f)
|
||||
files.add(f)
|
||||
|
||||
if files:
|
||||
return [
|
||||
output_api.PresubmitError(
|
||||
'Please avoid to use namespace `google::protobuf` directly.\n'
|
||||
'Add a using directive in `%s` and include that header instead.'
|
||||
% proto_utils_path, files)
|
||||
% proto_utils_path, list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -1117,19 +1118,19 @@ def CheckBannedAbslMakeUnique(input_api, output_api, source_file_filter):
|
|||
file_filter = lambda f: (f.LocalPath().endswith(
|
||||
('.cc', '.h')) and source_file_filter(f))
|
||||
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedFiles(include_deletes=False,
|
||||
file_filter=file_filter):
|
||||
for _, line in f.ChangedContents():
|
||||
if 'absl::make_unique' in line:
|
||||
files.append(f)
|
||||
files.add(f)
|
||||
break
|
||||
|
||||
if files:
|
||||
return [
|
||||
output_api.PresubmitError(
|
||||
'Please use std::make_unique instead of absl::make_unique.\n'
|
||||
'Affected files:', files)
|
||||
'Affected files:', list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -1142,20 +1143,20 @@ def CheckBannedAbslOptional(input_api, output_api, source_file_filter):
|
|||
file_filter = lambda f: (f.LocalPath().endswith(
|
||||
('.cc', '.h')) and source_file_filter(f))
|
||||
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedFiles(include_deletes=False,
|
||||
file_filter=file_filter):
|
||||
for _, line in f.ChangedContents():
|
||||
if absl_optional.search(line) or absl_optional_include.search(
|
||||
line):
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
break
|
||||
|
||||
if files:
|
||||
return [
|
||||
output_api.PresubmitError(
|
||||
'Please use std::optional instead of absl::optional.\n'
|
||||
'Affected files:', files)
|
||||
'Affected files:', list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -1171,19 +1172,19 @@ def CheckConditionalIncludes(input_api, output_api, source_file_filter):
|
|||
for key, value in conditional_includes.items():
|
||||
include_regex = re.compile('^#include ' + key +
|
||||
'((?!IWYU pragma|no-presubmit-check).)*$')
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedFiles(include_deletes=False,
|
||||
file_filter=file_filter):
|
||||
for _, line in f.ChangedContents():
|
||||
if include_regex.search(line):
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
break
|
||||
|
||||
if files:
|
||||
results.append(
|
||||
output_api.PresubmitError(
|
||||
'Please include ' + value + ' instead of ' + key +
|
||||
'.\nAffected files:', files))
|
||||
'.\nAffected files:', list(files)))
|
||||
return results
|
||||
|
||||
|
||||
|
|
@ -1193,7 +1194,7 @@ def CheckObjcApiSymbols(input_api, output_api, source_file_filter):
|
|||
file_filter = lambda f: (f.LocalPath().endswith(
|
||||
('.h')) and source_file_filter(f))
|
||||
|
||||
files = []
|
||||
files = set()
|
||||
file_filter = lambda x: (input_api.FilterSourceFile(x) and
|
||||
source_file_filter(x))
|
||||
for f in input_api.AffectedSourceFiles(file_filter):
|
||||
|
|
@ -1205,7 +1206,7 @@ def CheckObjcApiSymbols(input_api, output_api, source_file_filter):
|
|||
for match in rtc_objc_export.finditer(contents):
|
||||
export_block = match.group(0)
|
||||
if 'RTC_OBJC_TYPE' not in export_block:
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
|
||||
if len(files) > 0:
|
||||
return [
|
||||
|
|
@ -1214,7 +1215,7 @@ def CheckObjcApiSymbols(input_api, output_api, source_file_filter):
|
|||
+ 'macro.\n\n' + 'For example:\n' +
|
||||
'RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE(RtcFoo)\n\n' +
|
||||
'RTC_OBJC_EXPORT @interface RTC_OBJC_TYPE(RtcFoo)\n\n' +
|
||||
'Please fix the following files:', files)
|
||||
'Please fix the following files:', list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -1224,19 +1225,19 @@ def CheckAssertUsage(input_api, output_api, source_file_filter):
|
|||
file_filter = lambda f: (f.LocalPath().endswith(
|
||||
('.cc', '.h', '.m', '.mm')) and source_file_filter(f))
|
||||
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedFiles(include_deletes=False,
|
||||
file_filter=file_filter):
|
||||
for _, line in f.ChangedContents():
|
||||
if pattern.search(line):
|
||||
files.append(f.LocalPath())
|
||||
files.add(f.LocalPath())
|
||||
break
|
||||
|
||||
if len(files) > 0:
|
||||
return [
|
||||
output_api.PresubmitError(
|
||||
'Usage of assert() has been detected in the following files, '
|
||||
'please use RTC_DCHECK() instead.\n Files:', files)
|
||||
'please use RTC_DCHECK() instead.\n Files:', list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
@ -1247,7 +1248,7 @@ def CheckAbslMemoryInclude(input_api, output_api, source_file_filter):
|
|||
file_filter = lambda f: (f.LocalPath().endswith(
|
||||
('.cc', '.h')) and source_file_filter(f))
|
||||
|
||||
files = []
|
||||
files = set()
|
||||
for f in input_api.AffectedFiles(include_deletes=False,
|
||||
file_filter=file_filter):
|
||||
contents = input_api.ReadFile(f)
|
||||
|
|
@ -1255,7 +1256,7 @@ def CheckAbslMemoryInclude(input_api, output_api, source_file_filter):
|
|||
continue
|
||||
for _, line in f.ChangedContents():
|
||||
if 'absl::WrapUnique' in line:
|
||||
files.append(f)
|
||||
files.add(f)
|
||||
break
|
||||
|
||||
if len(files) > 0:
|
||||
|
|
@ -1263,7 +1264,8 @@ def CheckAbslMemoryInclude(input_api, output_api, source_file_filter):
|
|||
output_api.PresubmitError(
|
||||
'Please include "absl/memory/memory.h" header for '
|
||||
'absl::WrapUnique.\nThis header may or may not be included '
|
||||
'transitively depending on the C++ standard version.', files)
|
||||
'transitively depending on the C++ standard version.',
|
||||
list(files))
|
||||
]
|
||||
return []
|
||||
|
||||
|
|
|
|||
19
api/BUILD.gn
19
api/BUILD.gn
|
|
@ -42,6 +42,7 @@ rtc_source_set("enable_media") {
|
|||
"../pc:media_factory",
|
||||
"../rtc_base/system:rtc_export",
|
||||
"environment",
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -61,17 +62,12 @@ rtc_source_set("enable_media_with_defaults") {
|
|||
":libjingle_peerconnection_api",
|
||||
":scoped_refptr",
|
||||
"../rtc_base/system:rtc_export",
|
||||
"audio:audio_processing",
|
||||
"audio_codecs:audio_codecs_api",
|
||||
"audio_codecs:audio_codecs_api",
|
||||
"audio:builtin_audio_processing_builder",
|
||||
"audio_codecs:builtin_audio_decoder_factory",
|
||||
"audio_codecs:builtin_audio_encoder_factory",
|
||||
"task_queue:default_task_queue_factory",
|
||||
"task_queue:task_queue",
|
||||
"video_codecs:builtin_video_decoder_factory",
|
||||
"video_codecs:builtin_video_encoder_factory",
|
||||
"video_codecs:video_codecs_api",
|
||||
"video_codecs:video_codecs_api",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -98,6 +94,7 @@ if (!build_with_chromium) {
|
|||
"audio:audio_device",
|
||||
"audio:audio_mixer_api",
|
||||
"audio:audio_processing",
|
||||
"audio:builtin_audio_processing_builder",
|
||||
"audio_codecs:audio_codecs_api",
|
||||
"video_codecs:video_codecs_api",
|
||||
]
|
||||
|
|
@ -468,6 +465,7 @@ rtc_source_set("ref_count") {
|
|||
rtc_source_set("scoped_refptr") {
|
||||
visibility = [ "*" ]
|
||||
sources = [ "scoped_refptr.h" ]
|
||||
deps = [ "//third_party/abseil-cpp/absl/base:nullability" ]
|
||||
}
|
||||
|
||||
rtc_source_set("make_ref_counted") {
|
||||
|
|
@ -477,6 +475,7 @@ rtc_source_set("make_ref_counted") {
|
|||
":ref_count",
|
||||
":scoped_refptr",
|
||||
"../rtc_base:refcount",
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -1021,6 +1020,8 @@ if (rtc_include_tests) {
|
|||
"../modules/audio_processing",
|
||||
"../modules/audio_processing:audioproc_f_impl",
|
||||
"audio:audio_processing",
|
||||
"audio:builtin_audio_processing_builder",
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
]
|
||||
}
|
||||
|
||||
|
|
@ -1542,6 +1543,7 @@ if (rtc_include_tests) {
|
|||
"candidate_unittest.cc",
|
||||
"field_trials_unittest.cc",
|
||||
"function_view_unittest.cc",
|
||||
"jsep_unittest.cc",
|
||||
"rtc_error_unittest.cc",
|
||||
"rtc_event_log_output_file_unittest.cc",
|
||||
"rtp_packet_info_unittest.cc",
|
||||
|
|
@ -1575,6 +1577,7 @@ if (rtc_include_tests) {
|
|||
"../rtc_base:buffer",
|
||||
"../rtc_base:checks",
|
||||
"../rtc_base:gunit_helpers",
|
||||
"../rtc_base:logging",
|
||||
"../rtc_base:macromagic",
|
||||
"../rtc_base:platform_thread",
|
||||
"../rtc_base:rtc_event",
|
||||
|
|
@ -1607,6 +1610,10 @@ if (rtc_include_tests) {
|
|||
"//third_party/abseil-cpp/absl/strings",
|
||||
"//third_party/abseil-cpp/absl/strings:string_view",
|
||||
]
|
||||
|
||||
if (rtc_use_h265) {
|
||||
deps += [ "video:rtp_video_frame_h265_assembler_unittests" ]
|
||||
}
|
||||
}
|
||||
|
||||
rtc_library("compile_all_headers") {
|
||||
|
|
|
|||
|
|
@ -97,7 +97,7 @@ class ArrayViewBase {
|
|||
static_assert(Size > 0, "ArrayView size must be variable or non-negative");
|
||||
|
||||
public:
|
||||
ArrayViewBase(T* data, size_t size) : data_(data) {}
|
||||
ArrayViewBase(T* data, size_t /* size */) : data_(data) {}
|
||||
|
||||
static constexpr size_t size() { return Size; }
|
||||
static constexpr bool empty() { return false; }
|
||||
|
|
@ -114,7 +114,7 @@ class ArrayViewBase {
|
|||
template <typename T>
|
||||
class ArrayViewBase<T, 0> {
|
||||
public:
|
||||
explicit ArrayViewBase(T* data, size_t size) {}
|
||||
explicit ArrayViewBase(T* /* data */, size_t /* size */) {}
|
||||
|
||||
static constexpr size_t size() { return 0; }
|
||||
static constexpr bool empty() { return true; }
|
||||
|
|
|
|||
|
|
@ -36,7 +36,7 @@ size_t Call(ArrayView<T> av) {
|
|||
}
|
||||
|
||||
template <typename T, size_t N>
|
||||
void CallFixed(ArrayView<T, N> av) {}
|
||||
void CallFixed(ArrayView<T, N> /* av */) {}
|
||||
|
||||
} // namespace
|
||||
|
||||
|
|
|
|||
|
|
@ -78,12 +78,33 @@ rtc_source_set("audio_processing") {
|
|||
"../../rtc_base/system:arch",
|
||||
"../../rtc_base/system:file_wrapper",
|
||||
"../../rtc_base/system:rtc_export",
|
||||
"../environment",
|
||||
"../task_queue",
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
"//third_party/abseil-cpp/absl/strings:string_view",
|
||||
]
|
||||
}
|
||||
|
||||
rtc_library("builtin_audio_processing_builder") {
|
||||
visibility = [ "*" ]
|
||||
configs += [ "../../modules/audio_processing:apm_debug_dump" ]
|
||||
sources = [
|
||||
"builtin_audio_processing_builder.cc",
|
||||
"builtin_audio_processing_builder.h",
|
||||
]
|
||||
deps = [
|
||||
":audio_processing",
|
||||
":echo_control",
|
||||
"..:make_ref_counted",
|
||||
"..:scoped_refptr",
|
||||
"../../modules/audio_processing",
|
||||
"../../rtc_base:logging",
|
||||
"../../rtc_base/system:rtc_export",
|
||||
"../environment",
|
||||
"//third_party/abseil-cpp/absl/base:nullability",
|
||||
]
|
||||
}
|
||||
|
||||
rtc_source_set("audio_processing_statistics") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
|
|
@ -142,3 +163,24 @@ rtc_source_set("echo_detector_creator") {
|
|||
"../../modules/audio_processing:residual_echo_detector",
|
||||
]
|
||||
}
|
||||
|
||||
if (rtc_include_tests) {
|
||||
rtc_library("audio_api_unittests") {
|
||||
testonly = true
|
||||
sources = [
|
||||
"audio_processing_unittest.cc",
|
||||
"builtin_audio_processing_builder_unittest.cc",
|
||||
]
|
||||
deps = [
|
||||
":audio_processing",
|
||||
":builtin_audio_processing_builder",
|
||||
"..:make_ref_counted",
|
||||
"..:scoped_refptr",
|
||||
"../../modules/audio_processing:mocks",
|
||||
"../../test:test_support",
|
||||
"../environment",
|
||||
"../environment:environment_factory",
|
||||
"test:audio_api_unittests",
|
||||
]
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -57,7 +57,7 @@ class AudioTransport {
|
|||
uint32_t currentMicLevel,
|
||||
bool keyPressed,
|
||||
uint32_t& newMicLevel,
|
||||
std::optional<int64_t> estimatedCaptureTimeNS) { // NOLINT
|
||||
std::optional<int64_t> /* estimatedCaptureTimeNS */) { // NOLINT
|
||||
// TODO(webrtc:13620) Make the default behaver of the new API to behave as
|
||||
// the old API. This can be pure virtual if all uses of the old API is
|
||||
// removed.
|
||||
|
|
|
|||
|
|
@ -9,8 +9,14 @@
|
|||
*/
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include <string>
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/strings/string_builder.h"
|
||||
|
||||
|
|
@ -52,7 +58,7 @@ std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
|
|||
constexpr int AudioProcessing::kNativeSampleRatesHz[];
|
||||
|
||||
void CustomProcessing::SetRuntimeSetting(
|
||||
AudioProcessing::RuntimeSetting setting) {}
|
||||
AudioProcessing::RuntimeSetting /* setting */) {}
|
||||
|
||||
bool Agc1Config::operator==(const Agc1Config& rhs) const {
|
||||
const auto& analog_lhs = analog_gain_controller;
|
||||
|
|
@ -208,4 +214,25 @@ std::string AudioProcessing::Config::ToString() const {
|
|||
return builder.str();
|
||||
}
|
||||
|
||||
absl::Nonnull<std::unique_ptr<AudioProcessingBuilderInterface>>
|
||||
CustomAudioProcessing(
|
||||
absl::Nonnull<scoped_refptr<AudioProcessing>> audio_processing) {
|
||||
class Builder : public AudioProcessingBuilderInterface {
|
||||
public:
|
||||
explicit Builder(absl::Nonnull<scoped_refptr<AudioProcessing>> ap)
|
||||
: ap_(std::move(ap)) {}
|
||||
|
||||
absl::Nullable<scoped_refptr<AudioProcessing>> Build(
|
||||
const Environment& /*env*/) override {
|
||||
return std::move(ap_);
|
||||
}
|
||||
|
||||
private:
|
||||
absl::Nonnull<scoped_refptr<AudioProcessing>> ap_;
|
||||
};
|
||||
|
||||
RTC_CHECK(audio_processing);
|
||||
return std::make_unique<Builder>(std::move(audio_processing));
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -33,6 +33,7 @@
|
|||
#include "api/array_view.h"
|
||||
#include "api/audio/audio_processing_statistics.h"
|
||||
#include "api/audio/echo_control.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/ref_count.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
|
|
@ -85,7 +86,6 @@ class EchoDetector;
|
|||
// float interfaces use deinterleaved data.
|
||||
//
|
||||
// Usage example, omitting error checking:
|
||||
// rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
|
||||
//
|
||||
// AudioProcessing::Config config;
|
||||
// config.echo_canceller.enabled = true;
|
||||
|
|
@ -101,7 +101,8 @@ class EchoDetector;
|
|||
//
|
||||
// config.high_pass_filter.enabled = true;
|
||||
//
|
||||
// apm->ApplyConfig(config)
|
||||
// scoped_refptr<AudioProcessing> apm =
|
||||
// BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment());
|
||||
//
|
||||
// // Start a voice call...
|
||||
//
|
||||
|
|
@ -743,6 +744,23 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
|
|||
static int GetFrameSize(int sample_rate_hz) { return sample_rate_hz / 100; }
|
||||
};
|
||||
|
||||
class AudioProcessingBuilderInterface {
|
||||
public:
|
||||
virtual ~AudioProcessingBuilderInterface() = default;
|
||||
|
||||
virtual absl::Nullable<scoped_refptr<AudioProcessing>> Build(
|
||||
const Environment& env) = 0;
|
||||
};
|
||||
|
||||
// Returns builder that returns the `audio_processing` ignoring the extra
|
||||
// construction parameter `env`.
|
||||
// nullptr `audio_processing` is not supported as in some scenarios that imply
|
||||
// no audio processing, while in others - default builtin audio processing.
|
||||
// Callers should be explicit which of these two behaviors they want.
|
||||
absl::Nonnull<std::unique_ptr<AudioProcessingBuilderInterface>>
|
||||
CustomAudioProcessing(
|
||||
absl::Nonnull<scoped_refptr<AudioProcessing>> audio_processing);
|
||||
|
||||
// Experimental interface for a custom analysis submodule.
|
||||
class CustomAudioAnalyzer {
|
||||
public:
|
||||
|
|
@ -772,7 +790,8 @@ class CustomProcessing {
|
|||
virtual ~CustomProcessing() {}
|
||||
};
|
||||
|
||||
class RTC_EXPORT AudioProcessingBuilder {
|
||||
// Use BuiltinAudioProcessingBuilder instead, see bugs.webrtc.org/369904700
|
||||
class RTC_EXPORT [[deprecated]] AudioProcessingBuilder {
|
||||
public:
|
||||
AudioProcessingBuilder();
|
||||
AudioProcessingBuilder(const AudioProcessingBuilder&) = delete;
|
||||
|
|
|
|||
47
api/audio/audio_processing_unittest.cc
Normal file
47
api/audio/audio_processing_unittest.cc
Normal file
|
|
@ -0,0 +1,47 @@
|
|||
/*
|
||||
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
|
||||
#include <memory>
|
||||
|
||||
#include "api/environment/environment_factory.h"
|
||||
#include "api/make_ref_counted.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/audio_processing/include/mock_audio_processing.h"
|
||||
#include "test/gmock.h"
|
||||
#include "test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
using ::testing::_;
|
||||
using ::testing::NotNull;
|
||||
|
||||
TEST(CustomAudioProcessingTest, ReturnsPassedAudioProcessing) {
|
||||
scoped_refptr<AudioProcessing> ap =
|
||||
make_ref_counted<test::MockAudioProcessing>();
|
||||
|
||||
std::unique_ptr<AudioProcessingBuilderInterface> builder =
|
||||
CustomAudioProcessing(ap);
|
||||
|
||||
ASSERT_THAT(builder, NotNull());
|
||||
EXPECT_EQ(builder->Build(CreateEnvironment()), ap);
|
||||
}
|
||||
|
||||
#if GTEST_HAS_DEATH_TEST
|
||||
TEST(CustomAudioProcessingTest, NullptrAudioProcessingIsUnsupported) {
|
||||
#pragma clang diagnostic push
|
||||
#pragma clang diagnostic ignored "-Wnonnull"
|
||||
EXPECT_DEATH(CustomAudioProcessing(nullptr), _);
|
||||
#pragma clang diagnostic pop
|
||||
}
|
||||
#endif
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -167,7 +167,7 @@ class DeinterleavedView {
|
|||
};
|
||||
|
||||
template <typename T>
|
||||
constexpr size_t NumChannels(const MonoView<T>& view) {
|
||||
constexpr size_t NumChannels(const MonoView<T>& /* view */) {
|
||||
return 1u;
|
||||
}
|
||||
|
||||
|
|
@ -182,22 +182,22 @@ size_t NumChannels(const DeinterleavedView<T>& view) {
|
|||
}
|
||||
|
||||
template <typename T>
|
||||
constexpr bool IsMono(const MonoView<T>& view) {
|
||||
constexpr bool IsMono(const MonoView<T>& /* view */) {
|
||||
return true;
|
||||
}
|
||||
|
||||
template <typename T>
|
||||
constexpr bool IsInterleavedView(const MonoView<T>& view) {
|
||||
constexpr bool IsInterleavedView(const MonoView<T>& /* view */) {
|
||||
return true;
|
||||
}
|
||||
|
||||
template <typename T>
|
||||
constexpr bool IsInterleavedView(const InterleavedView<T>& view) {
|
||||
constexpr bool IsInterleavedView(const InterleavedView<T>& /* view */) {
|
||||
return true;
|
||||
}
|
||||
|
||||
template <typename T>
|
||||
constexpr bool IsInterleavedView(const DeinterleavedView<const T>& view) {
|
||||
constexpr bool IsInterleavedView(const DeinterleavedView<const T>& /* view */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
|
|
|
|||
34
api/audio/builtin_audio_processing_builder.cc
Normal file
34
api/audio/builtin_audio_processing_builder.cc
Normal file
|
|
@ -0,0 +1,34 @@
|
|||
/*
|
||||
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/make_ref_counted.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "modules/audio_processing/audio_processing_impl.h"
|
||||
#include "rtc_base/logging.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
absl::Nullable<scoped_refptr<AudioProcessing>>
|
||||
BuiltinAudioProcessingBuilder::Build(const Environment& /*env*/) {
|
||||
// TODO: bugs.webrtc.org/369904700 - Pass `env` when AudioProcessingImpl gets
|
||||
// constructor that accepts it.
|
||||
return make_ref_counted<AudioProcessingImpl>(
|
||||
config_, std::move(capture_post_processing_),
|
||||
std::move(render_pre_processing_), std::move(echo_control_factory_),
|
||||
std::move(echo_detector_), std::move(capture_analyzer_));
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -1,5 +1,5 @@
|
|||
/*
|
||||
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
|
||||
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
|
|
@ -8,65 +8,69 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
|
||||
#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
|
||||
#ifndef API_AUDIO_BUILTIN_AUDIO_PROCESSING_BUILDER_H_
|
||||
#define API_AUDIO_BUILTIN_AUDIO_PROCESSING_BUILDER_H_
|
||||
|
||||
#include <list>
|
||||
#include <memory>
|
||||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/echo_control.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "rtc_base/system/rtc_export.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
// Facilitates building of AudioProcessingImp for the tests.
|
||||
class AudioProcessingBuilderForTesting {
|
||||
class RTC_EXPORT BuiltinAudioProcessingBuilder
|
||||
: public AudioProcessingBuilderInterface {
|
||||
public:
|
||||
AudioProcessingBuilderForTesting();
|
||||
AudioProcessingBuilderForTesting(const AudioProcessingBuilderForTesting&) =
|
||||
delete;
|
||||
AudioProcessingBuilderForTesting& operator=(
|
||||
const AudioProcessingBuilderForTesting&) = delete;
|
||||
~AudioProcessingBuilderForTesting();
|
||||
BuiltinAudioProcessingBuilder() = default;
|
||||
explicit BuiltinAudioProcessingBuilder(const AudioProcessing::Config& config)
|
||||
: config_(config) {}
|
||||
BuiltinAudioProcessingBuilder(const BuiltinAudioProcessingBuilder&) = delete;
|
||||
BuiltinAudioProcessingBuilder& operator=(
|
||||
const BuiltinAudioProcessingBuilder&) = delete;
|
||||
~BuiltinAudioProcessingBuilder() override = default;
|
||||
|
||||
// Sets the APM configuration.
|
||||
AudioProcessingBuilderForTesting& SetConfig(
|
||||
BuiltinAudioProcessingBuilder& SetConfig(
|
||||
const AudioProcessing::Config& config) {
|
||||
config_ = config;
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo controller factory to inject when APM is created.
|
||||
AudioProcessingBuilderForTesting& SetEchoControlFactory(
|
||||
BuiltinAudioProcessingBuilder& SetEchoControlFactory(
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory) {
|
||||
echo_control_factory_ = std::move(echo_control_factory);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture post-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilderForTesting& SetCapturePostProcessing(
|
||||
BuiltinAudioProcessingBuilder& SetCapturePostProcessing(
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing) {
|
||||
capture_post_processing_ = std::move(capture_post_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the render pre-processing sub-module to inject when APM is created.
|
||||
AudioProcessingBuilderForTesting& SetRenderPreProcessing(
|
||||
BuiltinAudioProcessingBuilder& SetRenderPreProcessing(
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing) {
|
||||
render_pre_processing_ = std::move(render_pre_processing);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the echo detector to inject when APM is created.
|
||||
AudioProcessingBuilderForTesting& SetEchoDetector(
|
||||
BuiltinAudioProcessingBuilder& SetEchoDetector(
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector) {
|
||||
echo_detector_ = std::move(echo_detector);
|
||||
return *this;
|
||||
}
|
||||
|
||||
// Sets the capture analyzer sub-module to inject when APM is created.
|
||||
AudioProcessingBuilderForTesting& SetCaptureAnalyzer(
|
||||
BuiltinAudioProcessingBuilder& SetCaptureAnalyzer(
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer) {
|
||||
capture_analyzer_ = std::move(capture_analyzer);
|
||||
return *this;
|
||||
|
|
@ -74,22 +78,19 @@ class AudioProcessingBuilderForTesting {
|
|||
|
||||
// Creates an APM instance with the specified config or the default one if
|
||||
// unspecified. Injects the specified components transferring the ownership
|
||||
// to the newly created APM instance - i.e., except for the config, the
|
||||
// builder is reset to its initial state.
|
||||
rtc::scoped_refptr<AudioProcessing> Create();
|
||||
// to the newly created APM instance.
|
||||
absl::Nullable<scoped_refptr<AudioProcessing>> Build(
|
||||
const Environment& env) override;
|
||||
|
||||
private:
|
||||
// Transfers the ownership to a non-testing builder.
|
||||
void TransferOwnershipsToBuilder(AudioProcessingBuilder* builder);
|
||||
|
||||
AudioProcessing::Config config_;
|
||||
std::unique_ptr<EchoControlFactory> echo_control_factory_;
|
||||
std::unique_ptr<CustomProcessing> capture_post_processing_;
|
||||
std::unique_ptr<CustomProcessing> render_pre_processing_;
|
||||
rtc::scoped_refptr<EchoDetector> echo_detector_;
|
||||
scoped_refptr<EchoDetector> echo_detector_;
|
||||
std::unique_ptr<CustomAudioAnalyzer> capture_analyzer_;
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // MODULES_AUDIO_PROCESSING_TEST_AUDIO_PROCESSING_BUILDER_FOR_TESTING_H_
|
||||
#endif // API_AUDIO_BUILTIN_AUDIO_PROCESSING_BUILDER_H_
|
||||
48
api/audio/builtin_audio_processing_builder_unittest.cc
Normal file
48
api/audio/builtin_audio_processing_builder_unittest.cc
Normal file
|
|
@ -0,0 +1,48 @@
|
|||
/*
|
||||
* Copyright (c) 2024 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/environment/environment_factory.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "test/gmock.h"
|
||||
#include "test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
using ::testing::NotNull;
|
||||
|
||||
TEST(BuiltinAudioProcessingBuilderTest, CreatesWithDefaults) {
|
||||
EXPECT_THAT(BuiltinAudioProcessingBuilder().Build(CreateEnvironment()),
|
||||
NotNull());
|
||||
}
|
||||
|
||||
TEST(BuiltinAudioProcessingBuilderTest, CreatesWithConfig) {
|
||||
const Environment env = CreateEnvironment();
|
||||
AudioProcessing::Config config;
|
||||
// Change a field to make config different to default one.
|
||||
config.gain_controller1.enabled = !config.gain_controller1.enabled;
|
||||
|
||||
scoped_refptr<AudioProcessing> ap1 =
|
||||
BuiltinAudioProcessingBuilder(config).Build(env);
|
||||
ASSERT_THAT(ap1, NotNull());
|
||||
EXPECT_EQ(ap1->GetConfig().gain_controller1.enabled,
|
||||
config.gain_controller1.enabled);
|
||||
|
||||
scoped_refptr<AudioProcessing> ap2 =
|
||||
BuiltinAudioProcessingBuilder().SetConfig(config).Build(env);
|
||||
ASSERT_THAT(ap2, NotNull());
|
||||
EXPECT_EQ(ap2->GetConfig().gain_controller1.enabled,
|
||||
config.gain_controller1.enabled);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -53,7 +53,7 @@ class EchoControl {
|
|||
// resulting output is anyway not used, for instance when the endpoint is
|
||||
// muted.
|
||||
// TODO(b/177830919): Make pure virtual.
|
||||
virtual void SetCaptureOutputUsage(bool capture_output_used) {}
|
||||
virtual void SetCaptureOutputUsage(bool /* capture_output_used */) {}
|
||||
|
||||
// Returns wheter the signal is altered.
|
||||
virtual bool ActiveProcessing() const = 0;
|
||||
|
|
|
|||
|
|
@ -17,9 +17,9 @@
|
|||
namespace webrtc {
|
||||
|
||||
// Returns an instance of the WebRTC implementation of a residual echo detector.
|
||||
// It can be provided to the webrtc::AudioProcessingBuilder to obtain the
|
||||
// It can be provided to the webrtc::BuiltinAudioProcessingBuilder to obtain the
|
||||
// usual residual echo metrics.
|
||||
rtc::scoped_refptr<EchoDetector> CreateEchoDetector();
|
||||
scoped_refptr<EchoDetector> CreateEchoDetector();
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
|
|
|
|||
|
|
@ -66,12 +66,6 @@ rtc_library("builtin_audio_decoder_factory") {
|
|||
"g722:audio_decoder_g722",
|
||||
]
|
||||
defines = []
|
||||
if (rtc_include_ilbc) {
|
||||
deps += [ "ilbc:audio_decoder_ilbc" ]
|
||||
defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
|
||||
} else {
|
||||
defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
|
||||
}
|
||||
if (rtc_include_opus) {
|
||||
deps += [
|
||||
"opus:audio_decoder_multiopus",
|
||||
|
|
@ -99,15 +93,6 @@ rtc_library("builtin_audio_encoder_factory") {
|
|||
"g722:audio_encoder_g722",
|
||||
]
|
||||
defines = []
|
||||
if (rtc_include_ilbc) {
|
||||
deps += [
|
||||
"..:field_trials_view",
|
||||
"ilbc:audio_encoder_ilbc",
|
||||
]
|
||||
defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
|
||||
} else {
|
||||
defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
|
||||
}
|
||||
if (rtc_include_opus) {
|
||||
deps += [
|
||||
"..:field_trials_view",
|
||||
|
|
|
|||
|
|
@ -45,7 +45,7 @@ void AudioDecoderL16::AppendSupportedDecoders(
|
|||
std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
|
||||
const Config& config,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
if (!config.IsOk()) {
|
||||
return nullptr;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -72,7 +72,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
|
|||
const AudioEncoderL16::Config& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
AudioEncoderPcm16B::Config c;
|
||||
c.sample_rate_hz = config.sample_rate_hz;
|
||||
c.num_channels = config.num_channels;
|
||||
|
|
|
|||
|
|
@ -130,7 +130,8 @@ bool AudioDecoder::HasDecodePlc() const {
|
|||
return false;
|
||||
}
|
||||
|
||||
size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
|
||||
size_t AudioDecoder::DecodePlc(size_t /* num_frames */,
|
||||
int16_t* /* decoded */) {
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
@ -142,18 +143,18 @@ int AudioDecoder::ErrorCode() {
|
|||
return 0;
|
||||
}
|
||||
|
||||
int AudioDecoder::PacketDuration(const uint8_t* encoded,
|
||||
size_t encoded_len) const {
|
||||
int AudioDecoder::PacketDuration(const uint8_t* /* encoded */,
|
||||
size_t /* encoded_len */) const {
|
||||
return kNotImplemented;
|
||||
}
|
||||
|
||||
int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
|
||||
size_t encoded_len) const {
|
||||
int AudioDecoder::PacketDurationRedundant(const uint8_t* /* encoded */,
|
||||
size_t /* encoded_len */) const {
|
||||
return kNotImplemented;
|
||||
}
|
||||
|
||||
bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
|
||||
size_t encoded_len) const {
|
||||
bool AudioDecoder::PacketHasFec(const uint8_t* /* encoded */,
|
||||
size_t /* encoded_len */) const {
|
||||
return false;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -34,13 +34,16 @@ struct Helper;
|
|||
// Base case: 0 template parameters.
|
||||
template <>
|
||||
struct Helper<> {
|
||||
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {}
|
||||
static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
|
||||
static void AppendSupportedDecoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {}
|
||||
static bool IsSupportedDecoder(const SdpAudioFormat& /* format */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
static absl::Nullable<std::unique_ptr<AudioDecoder>> MakeAudioDecoder(
|
||||
const Environment& env,
|
||||
const SdpAudioFormat& format,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
const Environment& /* env */,
|
||||
const SdpAudioFormat& /* format */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
return nullptr;
|
||||
}
|
||||
};
|
||||
|
|
@ -72,7 +75,7 @@ template <typename Trait,
|
|||
std::unique_ptr<AudioDecoder>>>>
|
||||
absl::Nullable<std::unique_ptr<AudioDecoder>> CreateDecoder(
|
||||
Rank0,
|
||||
const Environment& env,
|
||||
const Environment& /* env */,
|
||||
const typename Trait::Config& config,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
return Trait::MakeAudioDecoder(config, codec_pair_id);
|
||||
|
|
|
|||
|
|
@ -67,31 +67,32 @@ bool AudioEncoder::GetDtx() const {
|
|||
return false;
|
||||
}
|
||||
|
||||
bool AudioEncoder::SetApplication(Application application) {
|
||||
bool AudioEncoder::SetApplication(Application /* application */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
|
||||
void AudioEncoder::SetMaxPlaybackRate(int /* frequency_hz */) {}
|
||||
|
||||
void AudioEncoder::SetTargetBitrate(int target_bps) {}
|
||||
void AudioEncoder::SetTargetBitrate(int /* target_bps */) {}
|
||||
|
||||
rtc::ArrayView<std::unique_ptr<AudioEncoder>>
|
||||
AudioEncoder::ReclaimContainedEncoders() {
|
||||
return nullptr;
|
||||
}
|
||||
|
||||
bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
|
||||
RtcEventLog* event_log) {
|
||||
bool AudioEncoder::EnableAudioNetworkAdaptor(
|
||||
const std::string& /* config_string */,
|
||||
RtcEventLog* /* event_log */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
void AudioEncoder::DisableAudioNetworkAdaptor() {}
|
||||
|
||||
void AudioEncoder::OnReceivedUplinkPacketLossFraction(
|
||||
float uplink_packet_loss_fraction) {}
|
||||
float /* uplink_packet_loss_fraction */) {}
|
||||
|
||||
void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
|
||||
float uplink_recoverable_packet_loss_fraction) {
|
||||
float /* uplink_recoverable_packet_loss_fraction */) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
}
|
||||
|
||||
|
|
@ -100,20 +101,20 @@ void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
|
|||
}
|
||||
|
||||
void AudioEncoder::OnReceivedUplinkBandwidth(
|
||||
int target_audio_bitrate_bps,
|
||||
std::optional<int64_t> bwe_period_ms) {}
|
||||
int /* target_audio_bitrate_bps */,
|
||||
std::optional<int64_t> /* bwe_period_ms */) {}
|
||||
|
||||
void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
|
||||
OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
|
||||
update.bwe_period.ms());
|
||||
}
|
||||
|
||||
void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
|
||||
void AudioEncoder::OnReceivedRtt(int /* rtt_ms */) {}
|
||||
|
||||
void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
|
||||
void AudioEncoder::OnReceivedOverhead(size_t /* overhead_bytes_per_packet */) {}
|
||||
|
||||
void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
|
||||
int max_frame_length_ms) {}
|
||||
void AudioEncoder::SetReceiverFrameLengthRange(int /* min_frame_length_ms */,
|
||||
int /* max_frame_length_ms */) {}
|
||||
|
||||
ANAStats AudioEncoder::GetANAStats() const {
|
||||
return ANAStats();
|
||||
|
|
|
|||
|
|
@ -35,15 +35,16 @@ struct Helper;
|
|||
// Base case: 0 template parameters.
|
||||
template <>
|
||||
struct Helper<> {
|
||||
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
|
||||
static void AppendSupportedEncoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {}
|
||||
static std::optional<AudioCodecInfo> QueryAudioEncoder(
|
||||
const SdpAudioFormat& format) {
|
||||
const SdpAudioFormat& /* format */) {
|
||||
return std::nullopt;
|
||||
}
|
||||
static absl::Nullable<std::unique_ptr<AudioEncoder>> CreateAudioEncoder(
|
||||
const Environment& env,
|
||||
const SdpAudioFormat& format,
|
||||
const AudioEncoderFactory::Options& options) {
|
||||
const Environment& /* env */,
|
||||
const SdpAudioFormat& /* format */,
|
||||
const AudioEncoderFactory::Options& /* options */) {
|
||||
return nullptr;
|
||||
}
|
||||
};
|
||||
|
|
@ -76,7 +77,7 @@ template <typename Trait,
|
|||
std::unique_ptr<AudioEncoder>>>>
|
||||
absl::Nullable<std::unique_ptr<AudioEncoder>> CreateEncoder(
|
||||
Rank0,
|
||||
const Environment& env,
|
||||
const Environment& /* env */,
|
||||
const typename Trait::Config& config,
|
||||
const AudioEncoderFactory::Options& options) {
|
||||
return Trait::MakeAudioEncoder(config, options.payload_type,
|
||||
|
|
|
|||
|
|
@ -23,9 +23,6 @@
|
|||
#include "api/audio_codecs/g711/audio_decoder_g711.h"
|
||||
#include "api/audio_codecs/g722/audio_decoder_g722.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#if WEBRTC_USE_BUILTIN_ILBC
|
||||
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
|
||||
#endif
|
||||
#if WEBRTC_USE_BUILTIN_OPUS
|
||||
#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
|
||||
#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
|
||||
|
|
@ -42,7 +39,8 @@ struct NotAdvertised {
|
|||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
return T::SdpToConfig(audio_format);
|
||||
}
|
||||
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
|
||||
static void AppendSupportedDecoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {
|
||||
// Don't advertise support for anything.
|
||||
}
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
|
|
@ -62,11 +60,6 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
|
|||
#endif
|
||||
|
||||
AudioDecoderG722,
|
||||
|
||||
#if WEBRTC_USE_BUILTIN_ILBC
|
||||
AudioDecoderIlbc,
|
||||
#endif
|
||||
|
||||
AudioDecoderG711, NotAdvertised<AudioDecoderL16>>();
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -24,9 +24,6 @@
|
|||
#include "api/audio_codecs/g722/audio_encoder_g722.h"
|
||||
#include "api/field_trials_view.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#if WEBRTC_USE_BUILTIN_ILBC
|
||||
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck
|
||||
#endif
|
||||
#if WEBRTC_USE_BUILTIN_OPUS
|
||||
#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
|
||||
#include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck
|
||||
|
|
@ -43,7 +40,8 @@ struct NotAdvertised {
|
|||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
return T::SdpToConfig(audio_format);
|
||||
}
|
||||
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
|
||||
static void AppendSupportedEncoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {
|
||||
// Don't advertise support for anything.
|
||||
}
|
||||
static AudioCodecInfo QueryAudioEncoder(const Config& config) {
|
||||
|
|
@ -69,11 +67,6 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
|
|||
#endif
|
||||
|
||||
AudioEncoderG722,
|
||||
|
||||
#if WEBRTC_USE_BUILTIN_ILBC
|
||||
AudioEncoderIlbc,
|
||||
#endif
|
||||
|
||||
AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -56,7 +56,7 @@ void AudioDecoderG711::AppendSupportedDecoders(
|
|||
std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
|
||||
const Config& config,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return nullptr;
|
||||
|
|
|
|||
|
|
@ -77,7 +77,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
|
|||
const Config& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return nullptr;
|
||||
|
|
|
|||
|
|
@ -44,7 +44,7 @@ void AudioDecoderG722::AppendSupportedDecoders(
|
|||
std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
|
||||
Config config,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return nullptr;
|
||||
|
|
|
|||
|
|
@ -75,7 +75,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
|
|||
const AudioEncoderG722Config& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return nullptr;
|
||||
|
|
|
|||
|
|
@ -1,53 +0,0 @@
|
|||
# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
#
|
||||
# Use of this source code is governed by a BSD-style license
|
||||
# that can be found in the LICENSE file in the root of the source
|
||||
# tree. An additional intellectual property rights grant can be found
|
||||
# in the file PATENTS. All contributing project authors may
|
||||
# be found in the AUTHORS file in the root of the source tree.
|
||||
|
||||
import("../../../webrtc.gni")
|
||||
if (is_android) {
|
||||
import("//build/config/android/config.gni")
|
||||
import("//build/config/android/rules.gni")
|
||||
}
|
||||
|
||||
rtc_source_set("audio_encoder_ilbc_config") {
|
||||
visibility = [ "*" ]
|
||||
sources = [ "audio_encoder_ilbc_config.h" ]
|
||||
}
|
||||
|
||||
rtc_library("audio_encoder_ilbc") {
|
||||
visibility = [ "*" ]
|
||||
poisonous = [ "audio_codecs" ]
|
||||
sources = [
|
||||
"audio_encoder_ilbc.cc",
|
||||
"audio_encoder_ilbc.h",
|
||||
]
|
||||
deps = [
|
||||
":audio_encoder_ilbc_config",
|
||||
"..:audio_codecs_api",
|
||||
"../../../api:field_trials_view",
|
||||
"../../../modules/audio_coding:ilbc",
|
||||
"../../../rtc_base:checks",
|
||||
"../../../rtc_base:safe_conversions",
|
||||
"../../../rtc_base:safe_minmax",
|
||||
"../../../rtc_base:stringutils",
|
||||
"//third_party/abseil-cpp/absl/strings",
|
||||
]
|
||||
}
|
||||
|
||||
rtc_library("audio_decoder_ilbc") {
|
||||
visibility = [ "*" ]
|
||||
poisonous = [ "audio_codecs" ]
|
||||
sources = [
|
||||
"audio_decoder_ilbc.cc",
|
||||
"audio_decoder_ilbc.h",
|
||||
]
|
||||
deps = [
|
||||
"..:audio_codecs_api",
|
||||
"../../../api:field_trials_view",
|
||||
"../../../modules/audio_coding:ilbc",
|
||||
"//third_party/abseil-cpp/absl/strings",
|
||||
]
|
||||
}
|
||||
|
|
@ -1,48 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
|
||||
|
||||
#include <memory>
|
||||
#include <optional>
|
||||
#include <vector>
|
||||
|
||||
#include "absl/strings/match.h"
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_decoder.h"
|
||||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/field_trials_view.h"
|
||||
#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
std::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
|
||||
const SdpAudioFormat& format) {
|
||||
if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
|
||||
format.clockrate_hz == 8000 && format.num_channels == 1) {
|
||||
return Config();
|
||||
}
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
void AudioDecoderIlbc::AppendSupportedDecoders(
|
||||
std::vector<AudioCodecSpec>* specs) {
|
||||
// RingRTC change to disable unused audio codecs
|
||||
// specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}});
|
||||
}
|
||||
|
||||
std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
|
||||
Config config,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
return std::make_unique<AudioDecoderIlbcImpl>();
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -1,39 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
|
||||
#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
|
||||
|
||||
#include <memory>
|
||||
#include <optional>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_decoder.h"
|
||||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/field_trials_view.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
// ILBC decoder API for use as a template parameter to
|
||||
// CreateAudioDecoderFactory<...>().
|
||||
struct AudioDecoderIlbc {
|
||||
struct Config {}; // Empty---no config values needed!
|
||||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
|
||||
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
Config config,
|
||||
std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
|
||||
const FieldTrialsView* field_trials = nullptr);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
|
||||
|
|
@ -1,98 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
|
||||
|
||||
#include <map>
|
||||
#include <memory>
|
||||
#include <optional>
|
||||
#include <string>
|
||||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "absl/strings/match.h"
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
|
||||
#include "api/field_trials_view.h"
|
||||
#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
|
||||
#include "rtc_base/checks.h"
|
||||
#include "rtc_base/numerics/safe_minmax.h"
|
||||
#include "rtc_base/string_to_number.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace {
|
||||
int GetIlbcBitrate(int ptime) {
|
||||
switch (ptime) {
|
||||
case 20:
|
||||
case 40:
|
||||
// 38 bytes per frame of 20 ms => 15200 bits/s.
|
||||
return 15200;
|
||||
case 30:
|
||||
case 60:
|
||||
// 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
|
||||
return 13333;
|
||||
default:
|
||||
RTC_CHECK_NOTREACHED();
|
||||
}
|
||||
}
|
||||
} // namespace
|
||||
|
||||
std::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
|
||||
const SdpAudioFormat& format) {
|
||||
if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
|
||||
format.clockrate_hz != 8000 || format.num_channels != 1) {
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
AudioEncoderIlbcConfig config;
|
||||
auto ptime_iter = format.parameters.find("ptime");
|
||||
if (ptime_iter != format.parameters.end()) {
|
||||
auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
|
||||
if (ptime && *ptime > 0) {
|
||||
const int whole_packets = *ptime / 10;
|
||||
config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
|
||||
}
|
||||
}
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return std::nullopt;
|
||||
}
|
||||
return config;
|
||||
}
|
||||
|
||||
void AudioEncoderIlbc::AppendSupportedEncoders(
|
||||
std::vector<AudioCodecSpec>* specs) {
|
||||
// RingRTC change to disable unused audio codecs
|
||||
// const SdpAudioFormat fmt = {"ILBC", 8000, 1};
|
||||
// const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
|
||||
// specs->push_back({fmt, info});
|
||||
}
|
||||
|
||||
AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
|
||||
const AudioEncoderIlbcConfig& config) {
|
||||
RTC_DCHECK(config.IsOk());
|
||||
return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
|
||||
}
|
||||
|
||||
std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
|
||||
const AudioEncoderIlbcConfig& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
if (!config.IsOk()) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return nullptr;
|
||||
}
|
||||
return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -1,43 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
|
||||
#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
|
||||
|
||||
#include <memory>
|
||||
#include <optional>
|
||||
#include <vector>
|
||||
|
||||
#include "api/audio_codecs/audio_codec_pair_id.h"
|
||||
#include "api/audio_codecs/audio_encoder.h"
|
||||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
|
||||
#include "api/field_trials_view.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
// ILBC encoder API for use as a template parameter to
|
||||
// CreateAudioEncoderFactory<...>().
|
||||
struct AudioEncoderIlbc {
|
||||
using Config = AudioEncoderIlbcConfig;
|
||||
static std::optional<AudioEncoderIlbcConfig> SdpToConfig(
|
||||
const SdpAudioFormat& audio_format);
|
||||
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
|
||||
static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const AudioEncoderIlbcConfig& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> codec_pair_id = std::nullopt,
|
||||
const FieldTrialsView* field_trials = nullptr);
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
|
||||
|
|
@ -1,28 +0,0 @@
|
|||
/*
|
||||
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
|
||||
#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
struct AudioEncoderIlbcConfig {
|
||||
bool IsOk() const {
|
||||
return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
|
||||
frame_size_ms == 60);
|
||||
}
|
||||
int frame_size_ms = 30; // Valid values are 20, 30, 40, and 60 ms.
|
||||
// Note that frame size 40 ms produces encodings with two 20 ms frames in
|
||||
// them, and frame size 60 ms consists of two 30 ms frames.
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
|
||||
|
|
@ -69,7 +69,7 @@ void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
|
|||
std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
|
||||
AudioDecoderMultiChannelOpusConfig config,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
|
||||
}
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -67,7 +67,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
|
|||
const AudioEncoderMultiChannelOpusConfig& config,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/,
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
|
||||
payload_type);
|
||||
}
|
||||
|
|
|
|||
|
|
@ -34,7 +34,8 @@ struct NotAdvertised {
|
|||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
return T::SdpToConfig(audio_format);
|
||||
}
|
||||
static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
|
||||
static void AppendSupportedDecoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {
|
||||
// Don't advertise support for anything.
|
||||
}
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
|
|
|
|||
|
|
@ -34,7 +34,8 @@ struct NotAdvertised {
|
|||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
return T::SdpToConfig(audio_format);
|
||||
}
|
||||
static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
|
||||
static void AppendSupportedEncoders(
|
||||
std::vector<AudioCodecSpec>* /* specs */) {
|
||||
// Don't advertise support for anything.
|
||||
}
|
||||
static AudioCodecInfo QueryAudioEncoder(const Config& config) {
|
||||
|
|
|
|||
|
|
@ -33,8 +33,6 @@ if (rtc_include_tests) {
|
|||
"../g711:audio_encoder_g711",
|
||||
"../g722:audio_decoder_g722",
|
||||
"../g722:audio_encoder_g722",
|
||||
"../ilbc:audio_decoder_ilbc",
|
||||
"../ilbc:audio_encoder_ilbc",
|
||||
"../opus:audio_decoder_opus",
|
||||
"../opus:audio_encoder_opus",
|
||||
]
|
||||
|
|
|
|||
|
|
@ -22,7 +22,6 @@
|
|||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/audio_codecs/g711/audio_decoder_g711.h"
|
||||
#include "api/audio_codecs/g722/audio_decoder_g722.h"
|
||||
#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
|
||||
#include "api/audio_codecs/opus/audio_decoder_opus.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/environment/environment_factory.h"
|
||||
|
|
@ -94,7 +93,8 @@ struct BaseAudioDecoderApi {
|
|||
|
||||
static SdpAudioFormat AudioFormat() { return {"fake", 16'000, 2, {}}; }
|
||||
|
||||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
static std::optional<Config> SdpToConfig(
|
||||
const SdpAudioFormat& /* audio_format */) {
|
||||
return Config();
|
||||
}
|
||||
|
||||
|
|
@ -111,17 +111,17 @@ struct TraitWithTwoMakeAudioDecoders : BaseAudioDecoderApi {
|
|||
static constexpr int kRateWithEnv = 20'000;
|
||||
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
const Config& config,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
const Config& /* config */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
auto decoder = std::make_unique<NiceMock<MockAudioDecoder>>();
|
||||
ON_CALL(*decoder, SampleRateHz).WillByDefault(Return(kRateWithoutEnv));
|
||||
return decoder;
|
||||
}
|
||||
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
const Environment& env,
|
||||
const Config& config,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
const Environment& /* env */,
|
||||
const Config& /* config */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
auto decoder = std::make_unique<NiceMock<MockAudioDecoder>>();
|
||||
ON_CALL(*decoder, SampleRateHz).WillByDefault(Return(kRateWithEnv));
|
||||
return decoder;
|
||||
|
|
@ -140,8 +140,8 @@ TEST(AudioDecoderFactoryTemplateTest,
|
|||
|
||||
struct AudioDecoderApiWithV1Make : BaseAudioDecoderApi {
|
||||
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||
const Config& config,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
const Config& /* config */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
return std::make_unique<NiceMock<MockAudioDecoder>>();
|
||||
}
|
||||
};
|
||||
|
|
@ -244,21 +244,6 @@ TEST(AudioDecoderFactoryTemplateTest, DISABLED_G722) {
|
|||
ASSERT_EQ(nullptr, dec3);
|
||||
}
|
||||
|
||||
// RingRTC change to disable unused audio codecs
|
||||
TEST(AudioDecoderFactoryTemplateTest, DISABLED_Ilbc) {
|
||||
const Environment env = CreateEnvironment();
|
||||
auto factory = CreateAudioDecoderFactory<AudioDecoderIlbc>();
|
||||
EXPECT_THAT(factory->GetSupportedDecoders(),
|
||||
::testing::ElementsAre(
|
||||
AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}}));
|
||||
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
|
||||
EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1}));
|
||||
EXPECT_EQ(nullptr, factory->Create(env, {"bar", 8000, 1}, std::nullopt));
|
||||
auto dec = factory->Create(env, {"ilbc", 8000, 1}, std::nullopt);
|
||||
ASSERT_NE(nullptr, dec);
|
||||
EXPECT_EQ(8000, dec->SampleRateHz());
|
||||
}
|
||||
|
||||
// RingRTC change to disable unused audio codecs
|
||||
TEST(AudioDecoderFactoryTemplateTest, DISABLED_L16) {
|
||||
const Environment env = CreateEnvironment();
|
||||
|
|
|
|||
|
|
@ -22,7 +22,6 @@
|
|||
#include "api/audio_codecs/audio_format.h"
|
||||
#include "api/audio_codecs/g711/audio_encoder_g711.h"
|
||||
#include "api/audio_codecs/g722/audio_encoder_g722.h"
|
||||
#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
|
||||
#include "api/audio_codecs/opus/audio_encoder_opus.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/environment/environment_factory.h"
|
||||
|
|
@ -78,7 +77,7 @@ struct AudioEncoderFakeApi {
|
|||
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const Config&,
|
||||
int payload_type,
|
||||
int /* payload_type */,
|
||||
std::optional<AudioCodecPairId> /*codec_pair_id*/ = std::nullopt) {
|
||||
auto enc = std::make_unique<testing::StrictMock<MockAudioEncoder>>();
|
||||
EXPECT_CALL(*enc, SampleRateHz())
|
||||
|
|
@ -100,7 +99,8 @@ struct BaseAudioEncoderApi {
|
|||
static SdpAudioFormat AudioFormat() { return {"fake", 16'000, 2, {}}; }
|
||||
static AudioCodecInfo CodecInfo() { return {16'000, 2, 23456}; }
|
||||
|
||||
static std::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
|
||||
static std::optional<Config> SdpToConfig(
|
||||
const SdpAudioFormat& /* audio_format */) {
|
||||
return Config();
|
||||
}
|
||||
|
||||
|
|
@ -114,8 +114,8 @@ struct BaseAudioEncoderApi {
|
|||
struct AudioEncoderApiWithV1Make : BaseAudioEncoderApi {
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const Config&,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
int /* payload_type */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
|
||||
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV1SameRate));
|
||||
return encoder;
|
||||
|
|
@ -124,9 +124,9 @@ struct AudioEncoderApiWithV1Make : BaseAudioEncoderApi {
|
|||
|
||||
struct AudioEncoderApiWithV2Make : BaseAudioEncoderApi {
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const Environment& env,
|
||||
const Config& config,
|
||||
const AudioEncoderFactory::Options& options) {
|
||||
const Environment& /* env */,
|
||||
const Config& /* config */,
|
||||
const AudioEncoderFactory::Options& /* options */) {
|
||||
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
|
||||
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV2SameRate));
|
||||
return encoder;
|
||||
|
|
@ -136,17 +136,17 @@ struct AudioEncoderApiWithV2Make : BaseAudioEncoderApi {
|
|||
struct AudioEncoderApiWithBothV1AndV2Make : BaseAudioEncoderApi {
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const Config&,
|
||||
int payload_type,
|
||||
std::optional<AudioCodecPairId> codec_pair_id) {
|
||||
int /* payload_type */,
|
||||
std::optional<AudioCodecPairId> /* codec_pair_id */) {
|
||||
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
|
||||
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV1SameRate));
|
||||
return encoder;
|
||||
}
|
||||
|
||||
static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
|
||||
const Environment& env,
|
||||
const Config& config,
|
||||
const AudioEncoderFactory::Options& options) {
|
||||
const Environment& /* env */,
|
||||
const Config& /* config */,
|
||||
const AudioEncoderFactory::Options& /* options */) {
|
||||
auto encoder = std::make_unique<NiceMock<MockAudioEncoder>>();
|
||||
ON_CALL(*encoder, SampleRateHz).WillByDefault(Return(kV2SameRate));
|
||||
return encoder;
|
||||
|
|
@ -268,22 +268,6 @@ TEST(AudioEncoderFactoryTemplateTest, DISABLED_G722) {
|
|||
Pointer(Property(&AudioEncoder::SampleRateHz, 16000)));
|
||||
}
|
||||
|
||||
// RingRTC change to disable unused audio codecs
|
||||
TEST(AudioEncoderFactoryTemplateTest, DISABLED_Ilbc) {
|
||||
const Environment env = CreateEnvironment();
|
||||
auto factory = CreateAudioEncoderFactory<AudioEncoderIlbc>();
|
||||
EXPECT_THAT(factory->GetSupportedEncoders(),
|
||||
::testing::ElementsAre(
|
||||
AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
|
||||
EXPECT_EQ(std::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
|
||||
EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
|
||||
factory->QueryAudioEncoder({"ilbc", 8000, 1}));
|
||||
|
||||
EXPECT_THAT(factory->Create(env, {"bar", 8000, 1}, {}), IsNull());
|
||||
EXPECT_THAT(factory->Create(env, {"ilbc", 8000, 1}, {}),
|
||||
Pointer(Property(&AudioEncoder::SampleRateHz, 8000)));
|
||||
}
|
||||
|
||||
// RingRTC change to disable unused audio codecs
|
||||
TEST(AudioEncoderFactoryTemplateTest, DISABLED_L16) {
|
||||
const Environment env = CreateEnvironment();
|
||||
|
|
|
|||
|
|
@ -10,6 +10,7 @@
|
|||
|
||||
#include "api/audio_options.h"
|
||||
|
||||
#include <optional>
|
||||
#include <string>
|
||||
|
||||
#include "api/array_view.h"
|
||||
|
|
|
|||
|
|
@ -27,8 +27,8 @@ struct PacketOptions {
|
|||
// Negative ids are invalid and should be interpreted
|
||||
// as packet_id not being set.
|
||||
int64_t packet_id = -1;
|
||||
// Whether this is a retransmission of an earlier packet.
|
||||
bool is_retransmit = false;
|
||||
// Whether this is an audio or video packet, excluding retransmissions.
|
||||
bool is_media = true;
|
||||
bool included_in_feedback = false;
|
||||
bool included_in_allocation = false;
|
||||
// Whether this packet can be part of a packet batch at lower levels.
|
||||
|
|
|
|||
|
|
@ -16,6 +16,7 @@
|
|||
#include "api/audio/audio_device.h"
|
||||
#include "api/audio/audio_mixer.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
#include "api/audio_codecs/audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/audio_encoder_factory.h"
|
||||
#include "api/enable_media.h"
|
||||
|
|
@ -57,10 +58,14 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
|
|||
dependencies.audio_encoder_factory = std::move(audio_encoder_factory);
|
||||
dependencies.audio_decoder_factory = std::move(audio_decoder_factory);
|
||||
dependencies.audio_frame_processor = std::move(audio_frame_processor);
|
||||
if (audio_processing) {
|
||||
dependencies.audio_processing = std::move(audio_processing);
|
||||
if (audio_processing != nullptr) {
|
||||
dependencies.audio_processing_builder =
|
||||
CustomAudioProcessing(std::move(audio_processing));
|
||||
} else {
|
||||
dependencies.audio_processing = AudioProcessingBuilder().Create();
|
||||
#ifndef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
|
||||
dependencies.audio_processing_builder =
|
||||
std::make_unique<BuiltinAudioProcessingBuilder>();
|
||||
#endif
|
||||
}
|
||||
dependencies.audio_mixer = std::move(audio_mixer);
|
||||
dependencies.video_encoder_factory = std::move(video_encoder_factory);
|
||||
|
|
|
|||
|
|
@ -11,6 +11,7 @@
|
|||
#include "api/data_channel_interface.h"
|
||||
|
||||
#include <cstdint>
|
||||
#include <optional>
|
||||
#include <string>
|
||||
|
||||
#include "absl/functional/any_invocable.h"
|
||||
|
|
@ -57,15 +58,15 @@ uint64_t DataChannelInterface::MaxSendQueueSize() {
|
|||
}
|
||||
|
||||
// TODO(tommi): Remove method once downstream implementations have been removed.
|
||||
bool DataChannelInterface::Send(const DataBuffer& buffer) {
|
||||
bool DataChannelInterface::Send(const DataBuffer& /* buffer */) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
return false;
|
||||
}
|
||||
|
||||
// TODO(tommi): Remove implementation once method is pure virtual.
|
||||
void DataChannelInterface::SendAsync(
|
||||
DataBuffer buffer,
|
||||
absl::AnyInvocable<void(RTCError) &&> on_complete) {
|
||||
DataBuffer /* buffer */,
|
||||
absl::AnyInvocable<void(RTCError) &&> /* on_complete */) {
|
||||
RTC_DCHECK_NOTREACHED();
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -99,7 +99,7 @@ class DataChannelObserver {
|
|||
// A data buffer was successfully received.
|
||||
virtual void OnMessage(const DataBuffer& buffer) = 0;
|
||||
// The data channel's buffered_amount has changed.
|
||||
virtual void OnBufferedAmountChange(uint64_t sent_data_size) {}
|
||||
virtual void OnBufferedAmountChange(uint64_t /* sent_data_size */) {}
|
||||
|
||||
// Override this to get callbacks directly on the network thread.
|
||||
// An implementation that does that must not block the network thread
|
||||
|
|
|
|||
|
|
@ -27,12 +27,12 @@ class DtmfSenderObserverInterface {
|
|||
// tones.
|
||||
// The callback includes the state of the tone buffer at the time when
|
||||
// the tone finished playing.
|
||||
virtual void OnToneChange(const std::string& tone,
|
||||
const std::string& tone_buffer) {}
|
||||
virtual void OnToneChange(const std::string& /* tone */,
|
||||
const std::string& /* tone_buffer */) {}
|
||||
// DEPRECATED: Older API without tone buffer.
|
||||
// TODO(bugs.webrtc.org/9725): Remove old API and default implementation
|
||||
// when old callers are gone.
|
||||
virtual void OnToneChange(const std::string& tone) {}
|
||||
virtual void OnToneChange(const std::string& /* tone */) {}
|
||||
|
||||
protected:
|
||||
virtual ~DtmfSenderObserverInterface() = default;
|
||||
|
|
@ -91,7 +91,7 @@ class DtmfSenderInterface : public webrtc::RefCountInterface {
|
|||
virtual bool InsertDtmf(const std::string& tones,
|
||||
int duration,
|
||||
int inter_tone_gap,
|
||||
int comma_delay) {
|
||||
int /* comma_delay */) {
|
||||
// TODO(bugs.webrtc.org/165700): Remove once downstream implementations
|
||||
// override this signature rather than the 3-parameter one.
|
||||
return InsertDtmf(tones, duration, inter_tone_gap);
|
||||
|
|
|
|||
|
|
@ -13,6 +13,7 @@
|
|||
#include <memory>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/peer_connection_interface.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
|
|
@ -45,11 +46,16 @@ class MediaFactoryImpl : public MediaFactory {
|
|||
std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
|
||||
const Environment& env,
|
||||
PeerConnectionFactoryDependencies& deps) override {
|
||||
absl::Nullable<scoped_refptr<AudioProcessing>> audio_processing =
|
||||
deps.audio_processing_builder != nullptr
|
||||
? std::move(deps.audio_processing_builder)->Build(env)
|
||||
: std::move(deps.audio_processing);
|
||||
|
||||
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
|
||||
&env.task_queue_factory(), deps.adm.get(),
|
||||
std::move(deps.audio_encoder_factory),
|
||||
std::move(deps.audio_decoder_factory), std::move(deps.audio_mixer),
|
||||
std::move(deps.audio_processing), std::move(deps.audio_frame_processor),
|
||||
std::move(audio_processing), std::move(deps.audio_frame_processor),
|
||||
env.field_trials());
|
||||
auto video_engine = std::make_unique<WebRtcVideoEngine>(
|
||||
std::move(deps.video_encoder_factory),
|
||||
|
|
|
|||
|
|
@ -12,7 +12,7 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
|
||||
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
|
||||
#include "api/enable_media.h"
|
||||
|
|
@ -34,10 +34,11 @@ void EnableMediaWithDefaults(PeerConnectionFactoryDependencies& deps) {
|
|||
if (deps.audio_decoder_factory == nullptr) {
|
||||
deps.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
|
||||
}
|
||||
if (deps.audio_processing == nullptr) {
|
||||
deps.audio_processing = AudioProcessingBuilder().Create();
|
||||
if (deps.audio_processing == nullptr &&
|
||||
deps.audio_processing_builder == nullptr) {
|
||||
deps.audio_processing_builder =
|
||||
std::make_unique<BuiltinAudioProcessingBuilder>();
|
||||
}
|
||||
|
||||
if (deps.video_encoder_factory == nullptr) {
|
||||
deps.video_encoder_factory = CreateBuiltinVideoEncoderFactory();
|
||||
}
|
||||
|
|
|
|||
|
|
@ -124,7 +124,7 @@ inline void EnvironmentFactory::Set(absl::Nullable<RtcEventLog*> utility) {
|
|||
|
||||
namespace webrtc_create_environment_internal {
|
||||
|
||||
inline void Set(EnvironmentFactory& factory) {}
|
||||
inline void Set(EnvironmentFactory& /* factory */) {}
|
||||
|
||||
template <typename FirstUtility, typename... Utilities>
|
||||
void Set(EnvironmentFactory& factory,
|
||||
|
|
|
|||
|
|
@ -54,7 +54,9 @@ class FakeFieldTrials : public FieldTrialsView {
|
|||
}
|
||||
}
|
||||
|
||||
std::string Lookup(absl::string_view key) const override { return "fake"; }
|
||||
std::string Lookup(absl::string_view /* key */) const override {
|
||||
return "fake";
|
||||
}
|
||||
|
||||
private:
|
||||
absl::AnyInvocable<void() &&> on_destroyed_;
|
||||
|
|
@ -72,8 +74,8 @@ class FakeTaskQueueFactory : public TaskQueueFactory {
|
|||
}
|
||||
|
||||
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
|
||||
absl::string_view name,
|
||||
Priority priority) const override {
|
||||
absl::string_view /* name */,
|
||||
Priority /* priority */) const override {
|
||||
return nullptr;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -54,9 +54,15 @@ class TransformableFrameInterface {
|
|||
virtual uint32_t GetTimestamp() const = 0;
|
||||
virtual void SetRTPTimestamp(uint32_t timestamp) = 0;
|
||||
|
||||
// TODO(https://bugs.webrtc.org/373365537): Remove this once its usage is
|
||||
// removed from blink.
|
||||
virtual std::optional<Timestamp> GetCaptureTimeIdentifier() const {
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
// TODO(https://bugs.webrtc.org/14878): Change this to pure virtual after it
|
||||
// is implemented everywhere.
|
||||
virtual std::optional<Timestamp> GetCaptureTimeIdentifier() const {
|
||||
virtual std::optional<Timestamp> GetPresentationTimestamp() const {
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
|
|
@ -139,9 +145,9 @@ class FrameTransformerInterface : public RefCountInterface {
|
|||
rtc::scoped_refptr<TransformedFrameCallback>) {}
|
||||
virtual void RegisterTransformedFrameSinkCallback(
|
||||
rtc::scoped_refptr<TransformedFrameCallback>,
|
||||
uint32_t ssrc) {}
|
||||
uint32_t /* ssrc */) {}
|
||||
virtual void UnregisterTransformedFrameCallback() {}
|
||||
virtual void UnregisterTransformedFrameSinkCallback(uint32_t ssrc) {}
|
||||
virtual void UnregisterTransformedFrameSinkCallback(uint32_t /* ssrc */) {}
|
||||
|
||||
protected:
|
||||
~FrameTransformerInterface() override = default;
|
||||
|
|
|
|||
|
|
@ -86,7 +86,7 @@ class FunctionView<RetT(ArgT...)> final {
|
|||
typename std::enable_if<std::is_same<
|
||||
std::nullptr_t,
|
||||
typename std::remove_cv<F>::type>::value>::type* = nullptr>
|
||||
FunctionView(F&& f) : call_(nullptr) {}
|
||||
FunctionView(F&& /* f */) : call_(nullptr) {}
|
||||
|
||||
// Default constructor. Creates an empty FunctionView.
|
||||
FunctionView() : call_(nullptr) {}
|
||||
|
|
|
|||
|
|
@ -11,6 +11,7 @@
|
|||
#include "api/jsep.h"
|
||||
|
||||
#include <cstddef>
|
||||
#include <optional>
|
||||
#include <string>
|
||||
#include <vector>
|
||||
|
||||
|
|
@ -21,7 +22,7 @@ std::string IceCandidateInterface::server_url() const {
|
|||
}
|
||||
|
||||
size_t SessionDescriptionInterface::RemoveCandidates(
|
||||
const std::vector<cricket::Candidate>& candidates) {
|
||||
const std::vector<cricket::Candidate>& /* candidates */) {
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
|
|
|||
13
api/jsep.h
13
api/jsep.h
|
|
@ -187,6 +187,19 @@ class RTC_EXPORT SessionDescriptionInterface {
|
|||
|
||||
// Serializes the description to SDP.
|
||||
virtual bool ToString(std::string* out) const = 0;
|
||||
template <typename Sink>
|
||||
friend void AbslStringify(Sink& sink, const SessionDescriptionInterface& p) {
|
||||
sink.Append("\n--- BEGIN SDP ");
|
||||
sink.Append(SdpTypeToString(p.GetType()));
|
||||
sink.Append(" ---\n");
|
||||
std::string temp;
|
||||
if (p.ToString(&temp)) {
|
||||
sink.Append(temp);
|
||||
} else {
|
||||
sink.Append("Error in ToString\n");
|
||||
}
|
||||
sink.Append("--- END SDP ---\n");
|
||||
}
|
||||
};
|
||||
|
||||
// Creates a SessionDescriptionInterface based on the SDP string and the type.
|
||||
|
|
|
|||
67
api/jsep_unittest.cc
Normal file
67
api/jsep_unittest.cc
Normal file
|
|
@ -0,0 +1,67 @@
|
|||
/*
|
||||
* Copyright 2024 The WebRTC project authors. All Rights Reserved.
|
||||
*
|
||||
* Use of this source code is governed by a BSD-style license
|
||||
* that can be found in the LICENSE file in the root of the source
|
||||
* tree. An additional intellectual property rights grant can be found
|
||||
* in the file PATENTS. All contributing project authors may
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "api/jsep.h"
|
||||
|
||||
#include <memory>
|
||||
#include <string>
|
||||
|
||||
#include "absl/strings/str_cat.h"
|
||||
#include "rtc_base/logging.h"
|
||||
#include "test/gmock.h"
|
||||
#include "test/gtest.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
using ::testing::HasSubstr;
|
||||
|
||||
TEST(JsepTest, AbslStringifySdp) {
|
||||
std::string sdp =
|
||||
"v=0\r\n"
|
||||
"o=- 0 3 IN IP4 127.0.0.1\r\n"
|
||||
"s=-\r\n"
|
||||
"t=0 0\r\n"
|
||||
"a=group:BUNDLE 0 1\r\n"
|
||||
"a=fingerprint:sha-1 "
|
||||
"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
|
||||
"a=setup:actpass\r\n"
|
||||
"a=ice-ufrag:ETEn\r\n"
|
||||
"a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"
|
||||
"m=audio 9 UDP/TLS/RTP/SAVPF 111\r\n"
|
||||
"c=IN IP4 0.0.0.0\r\n"
|
||||
"a=rtcp-mux\r\n"
|
||||
"a=sendonly\r\n"
|
||||
"a=mid:0\r\n"
|
||||
"a=rtpmap:111 opus/48000/2\r\n"
|
||||
"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
|
||||
"c=IN IP4 0.0.0.0\r\n"
|
||||
"a=rtcp-mux\r\n"
|
||||
"a=sendonly\r\n"
|
||||
"a=mid:1\r\n"
|
||||
"a=rtpmap:111 H264/90000\r\n"
|
||||
"a=fmtp:111 "
|
||||
"level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
|
||||
"42e01f\r\n";
|
||||
|
||||
std::unique_ptr<SessionDescriptionInterface> some_sdp =
|
||||
CreateSessionDescription(SdpType::kOffer, sdp);
|
||||
// Verify that sending the SDP to the log compiles.
|
||||
RTC_LOG(LS_VERBOSE) << "The SDP is " << *some_sdp;
|
||||
// Since create/stringify mangles order of fields, we only test
|
||||
// some substrings.
|
||||
EXPECT_THAT(absl::StrCat(*some_sdp), HasSubstr("a=rtpmap:111 opus/48000"));
|
||||
EXPECT_THAT(
|
||||
absl::StrCat(*some_sdp),
|
||||
HasSubstr(
|
||||
"a=fingerprint:sha-1 "
|
||||
"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"));
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
@ -13,6 +13,7 @@
|
|||
#include <type_traits>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/ref_count.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "rtc_base/ref_counted_object.h"
|
||||
|
|
@ -85,7 +86,7 @@ template <
|
|||
typename std::enable_if<std::is_convertible_v<T*, RefCountInterface*> &&
|
||||
std::is_abstract_v<T>,
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<T> make_ref_counted(Args&&... args) {
|
||||
absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
|
||||
return scoped_refptr<T>(new RefCountedObject<T>(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
|
|
@ -98,7 +99,7 @@ template <
|
|||
!std::is_convertible_v<T*, RefCountInterface*> &&
|
||||
webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<T> make_ref_counted(Args&&... args) {
|
||||
absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
|
||||
return scoped_refptr<T>(new T(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
|
|
@ -112,61 +113,18 @@ template <
|
|||
!webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
|
||||
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<FinalRefCountedObject<T>> make_ref_counted(Args&&... args) {
|
||||
absl::Nonnull<scoped_refptr<FinalRefCountedObject<T>>> make_ref_counted(
|
||||
Args&&... args) {
|
||||
return scoped_refptr<FinalRefCountedObject<T>>(
|
||||
new FinalRefCountedObject<T>(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
// Backwards compatibe aliases.
|
||||
// TODO: https://issues.webrtc.org/42225969 - deprecate and remove.
|
||||
namespace rtc {
|
||||
// This doesn't work:
|
||||
// template <typename T, typename... Args>
|
||||
// using make_ref_counted(Args&&... args) =
|
||||
// webrtc::make_ref_counted<T>(Args&&... args);
|
||||
// Instead, reproduce the templates.
|
||||
template <typename T,
|
||||
typename... Args,
|
||||
typename std::enable_if<
|
||||
std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
|
||||
std::is_abstract_v<T>,
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<T> make_ref_counted(Args&&... args) {
|
||||
return webrtc::scoped_refptr<T>(
|
||||
new webrtc::RefCountedObject<T>(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
// `make_ref_counted` for complete classes that are not convertible to
|
||||
// RefCountInterface and already carry a ref count.
|
||||
template <typename T,
|
||||
typename... Args,
|
||||
typename std::enable_if<
|
||||
!std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
|
||||
webrtc::webrtc_make_ref_counted_internal::HasAddRefAndRelease<
|
||||
T>::value,
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<T> make_ref_counted(Args&&... args) {
|
||||
return webrtc::scoped_refptr<T>(new T(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
// `make_ref_counted` for complete classes that are not convertible to
|
||||
// RefCountInterface and have no ref count of their own.
|
||||
template <typename T,
|
||||
typename... Args,
|
||||
typename std::enable_if<
|
||||
!std::is_convertible_v<T*, webrtc::RefCountInterface*> &&
|
||||
!webrtc::webrtc_make_ref_counted_internal::
|
||||
HasAddRefAndRelease<T>::value,
|
||||
|
||||
T>::type* = nullptr>
|
||||
scoped_refptr<webrtc::FinalRefCountedObject<T>> make_ref_counted(
|
||||
Args&&... args) {
|
||||
return webrtc::scoped_refptr<FinalRefCountedObject<T>>(
|
||||
new webrtc::FinalRefCountedObject<T>(std::forward<Args>(args)...));
|
||||
}
|
||||
|
||||
// Backwards compatibe alias.
|
||||
// TODO: bugs.webrtc.org/42225969 - deprecate and remove.
|
||||
using ::webrtc::make_ref_counted;
|
||||
} // namespace rtc
|
||||
|
||||
#endif // API_MAKE_REF_COUNTED_H_
|
||||
|
|
|
|||
|
|
@ -25,7 +25,7 @@ VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
|
|||
return ContentHint::kNone;
|
||||
}
|
||||
|
||||
bool AudioTrackInterface::GetSignalLevel(int* level) {
|
||||
bool AudioTrackInterface::GetSignalLevel(int* /* level */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -161,7 +161,7 @@ class VideoTrackSourceInterface : public MediaSourceInterface,
|
|||
// The call is expected to happen on the network thread.
|
||||
// TODO(crbug/1255737): make pure virtual once downstream project adapts.
|
||||
virtual void ProcessConstraints(
|
||||
const webrtc::VideoTrackSourceConstraints& constraints) {}
|
||||
const webrtc::VideoTrackSourceConstraints& /* constraints */) {}
|
||||
|
||||
protected:
|
||||
~VideoTrackSourceInterface() override = default;
|
||||
|
|
@ -184,14 +184,14 @@ class RTC_EXPORT VideoTrackInterface
|
|||
|
||||
// Register a video sink for this track. Used to connect the track to the
|
||||
// underlying video engine.
|
||||
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
|
||||
const rtc::VideoSinkWants& wants) override {}
|
||||
void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
|
||||
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* /* sink */,
|
||||
const rtc::VideoSinkWants& /* wants */) override {}
|
||||
void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* /* sink */) override {}
|
||||
|
||||
virtual VideoTrackSourceInterface* GetSource() const = 0;
|
||||
|
||||
virtual ContentHint content_hint() const;
|
||||
virtual void set_content_hint(ContentHint hint) {}
|
||||
virtual void set_content_hint(ContentHint /* hint */) {}
|
||||
|
||||
protected:
|
||||
~VideoTrackInterface() override = default;
|
||||
|
|
@ -200,11 +200,11 @@ class RTC_EXPORT VideoTrackInterface
|
|||
// Interface for receiving audio data from a AudioTrack.
|
||||
class AudioTrackSinkInterface {
|
||||
public:
|
||||
virtual void OnData(const void* audio_data,
|
||||
int bits_per_sample,
|
||||
int sample_rate,
|
||||
size_t number_of_channels,
|
||||
size_t number_of_frames) {
|
||||
virtual void OnData(const void* /* audio_data */,
|
||||
int /* bits_per_sample */,
|
||||
int /* sample_rate */,
|
||||
size_t /* number_of_channels */,
|
||||
size_t /* number_of_frames */) {
|
||||
RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
|
||||
}
|
||||
|
||||
|
|
@ -212,12 +212,13 @@ class AudioTrackSinkInterface {
|
|||
// supposed to deliver the timestamp when this audio frame was originally
|
||||
// captured. This timestamp MUST be based on the same clock as
|
||||
// rtc::TimeMillis().
|
||||
virtual void OnData(const void* audio_data,
|
||||
int bits_per_sample,
|
||||
int sample_rate,
|
||||
size_t number_of_channels,
|
||||
size_t number_of_frames,
|
||||
std::optional<int64_t> absolute_capture_timestamp_ms) {
|
||||
virtual void OnData(
|
||||
const void* audio_data,
|
||||
int bits_per_sample,
|
||||
int sample_rate,
|
||||
size_t number_of_channels,
|
||||
size_t number_of_frames,
|
||||
std::optional<int64_t> /* absolute_capture_timestamp_ms */) {
|
||||
// TODO(bugs.webrtc.org/10739): Deprecate the old OnData and make this one
|
||||
// pure virtual.
|
||||
return OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
|
||||
|
|
@ -251,15 +252,15 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
|
|||
// Sets the volume of the source. `volume` is in the range of [0, 10].
|
||||
// TODO(tommi): This method should be on the track and ideally volume should
|
||||
// be applied in the track in a way that does not affect clones of the track.
|
||||
virtual void SetVolume(double volume) {}
|
||||
virtual void SetVolume(double /* volume */) {}
|
||||
|
||||
// Registers/unregisters observers to the audio source.
|
||||
virtual void RegisterAudioObserver(AudioObserver* observer) {}
|
||||
virtual void UnregisterAudioObserver(AudioObserver* observer) {}
|
||||
virtual void RegisterAudioObserver(AudioObserver* /* observer */) {}
|
||||
virtual void UnregisterAudioObserver(AudioObserver* /* observer */) {}
|
||||
|
||||
// TODO(tommi): Make pure virtual.
|
||||
virtual void AddSink(AudioTrackSinkInterface* sink) {}
|
||||
virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
|
||||
virtual void AddSink(AudioTrackSinkInterface* /* sink */) {}
|
||||
virtual void RemoveSink(AudioTrackSinkInterface* /* sink */) {}
|
||||
|
||||
// Returns options for the AudioSource.
|
||||
// (for some of the settings this approach is broken, e.g. setting
|
||||
|
|
@ -339,35 +340,20 @@ class MediaStreamInterface : public webrtc::RefCountInterface,
|
|||
// Note: Default implementations are for avoiding link time errors in
|
||||
// implementations that mock this API.
|
||||
// TODO(bugs.webrtc.org/13980): Remove default implementations.
|
||||
virtual bool AddTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
|
||||
virtual bool AddTrack(rtc::scoped_refptr<AudioTrackInterface> /* track */) {
|
||||
RTC_CHECK_NOTREACHED();
|
||||
}
|
||||
virtual bool AddTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
|
||||
virtual bool AddTrack(rtc::scoped_refptr<VideoTrackInterface> /* track */) {
|
||||
RTC_CHECK_NOTREACHED();
|
||||
}
|
||||
virtual bool RemoveTrack(rtc::scoped_refptr<AudioTrackInterface> track) {
|
||||
virtual bool RemoveTrack(
|
||||
rtc::scoped_refptr<AudioTrackInterface> /* track */) {
|
||||
RTC_CHECK_NOTREACHED();
|
||||
}
|
||||
virtual bool RemoveTrack(rtc::scoped_refptr<VideoTrackInterface> track) {
|
||||
virtual bool RemoveTrack(
|
||||
rtc::scoped_refptr<VideoTrackInterface> /* track */) {
|
||||
RTC_CHECK_NOTREACHED();
|
||||
}
|
||||
// Deprecated: Should use scoped_refptr versions rather than pointers.
|
||||
[[deprecated("Pass a scoped_refptr")]] virtual bool AddTrack(
|
||||
AudioTrackInterface* track) {
|
||||
return AddTrack(rtc::scoped_refptr<AudioTrackInterface>(track));
|
||||
}
|
||||
[[deprecated("Pass a scoped_refptr")]] virtual bool AddTrack(
|
||||
VideoTrackInterface* track) {
|
||||
return AddTrack(rtc::scoped_refptr<VideoTrackInterface>(track));
|
||||
}
|
||||
[[deprecated("Pass a scoped_refptr")]] virtual bool RemoveTrack(
|
||||
AudioTrackInterface* track) {
|
||||
return RemoveTrack(rtc::scoped_refptr<AudioTrackInterface>(track));
|
||||
}
|
||||
[[deprecated("Pass a scoped_refptr")]] virtual bool RemoveTrack(
|
||||
VideoTrackInterface* track) {
|
||||
return RemoveTrack(rtc::scoped_refptr<VideoTrackInterface>(track));
|
||||
}
|
||||
|
||||
protected:
|
||||
~MediaStreamInterface() override = default;
|
||||
|
|
|
|||
|
|
@ -36,7 +36,8 @@ class RTC_EXPORT Metronome {
|
|||
// executed on the same sequence as they were requested on. There are no
|
||||
// features for cancellation. When that's needed, use e.g. ScopedTaskSafety
|
||||
// from the client.
|
||||
virtual void RequestCallOnNextTick(absl::AnyInvocable<void() &&> callback) {}
|
||||
virtual void RequestCallOnNextTick(
|
||||
absl::AnyInvocable<void() &&> /* callback */) {}
|
||||
|
||||
// Returns the current tick period of the metronome.
|
||||
virtual TimeDelta TickPeriod() const = 0;
|
||||
|
|
|
|||
|
|
@ -28,6 +28,22 @@ rtc_source_set("neteq_api") {
|
|||
]
|
||||
}
|
||||
|
||||
rtc_source_set("default_neteq_factory") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
"default_neteq_factory.cc",
|
||||
"default_neteq_factory.h",
|
||||
]
|
||||
deps = [
|
||||
":default_neteq_controller_factory",
|
||||
":neteq_api",
|
||||
"..:scoped_refptr",
|
||||
"../../modules/audio_coding:neteq",
|
||||
"../audio_codecs:audio_codecs_api",
|
||||
"../environment",
|
||||
]
|
||||
}
|
||||
|
||||
rtc_source_set("custom_neteq_factory") {
|
||||
visibility = [ "*" ]
|
||||
sources = [
|
||||
|
|
|
|||
|
|
@ -8,7 +8,7 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#include "modules/audio_coding/neteq/default_neteq_factory.h"
|
||||
#include "api/neteq/default_neteq_factory.h"
|
||||
|
||||
#include <utility>
|
||||
|
||||
|
|
@ -8,8 +8,8 @@
|
|||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
|
||||
#ifndef MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
#define MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
#ifndef API_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
#define API_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
|
||||
#include <memory>
|
||||
|
||||
|
|
@ -38,4 +38,4 @@ class DefaultNetEqFactory : public NetEqFactory {
|
|||
};
|
||||
|
||||
} // namespace webrtc
|
||||
#endif // MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
#endif // API_NETEQ_DEFAULT_NETEQ_FACTORY_H_
|
||||
|
|
@ -194,7 +194,7 @@ class NetEq {
|
|||
// Returns 0 on success, -1 on failure.
|
||||
virtual int InsertPacket(const RTPHeader& rtp_header,
|
||||
rtc::ArrayView<const uint8_t> payload,
|
||||
Timestamp receive_time) {
|
||||
Timestamp /* receive_time */) {
|
||||
// TODO: webrtc:343501093 - Make this method pure virtual.
|
||||
return InsertPacket(rtp_header, payload);
|
||||
}
|
||||
|
|
@ -300,7 +300,7 @@ class NetEq {
|
|||
// such payload type was registered.
|
||||
[[deprecated(
|
||||
"Use GetCurrentDecoderFormat")]] virtual std::optional<DecoderFormat>
|
||||
GetDecoderFormat(int payload_type) const {
|
||||
GetDecoderFormat(int /* payload_type */) const {
|
||||
return std::nullopt;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -147,18 +147,18 @@ PeerConnectionFactoryInterface::CreatePeerConnection(
|
|||
|
||||
RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
|
||||
PeerConnectionFactoryInterface::CreatePeerConnectionOrError(
|
||||
const PeerConnectionInterface::RTCConfiguration& configuration,
|
||||
PeerConnectionDependencies dependencies) {
|
||||
const PeerConnectionInterface::RTCConfiguration& /* configuration */,
|
||||
PeerConnectionDependencies /* dependencies */) {
|
||||
return RTCError(RTCErrorType::INTERNAL_ERROR);
|
||||
}
|
||||
|
||||
RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities(
|
||||
cricket::MediaType kind) const {
|
||||
cricket::MediaType /* kind */) const {
|
||||
return {};
|
||||
}
|
||||
|
||||
RtpCapabilities PeerConnectionFactoryInterface::GetRtpReceiverCapabilities(
|
||||
cricket::MediaType kind) const {
|
||||
cricket::MediaType /* kind */) const {
|
||||
return {};
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -360,6 +360,13 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
bool dscp() const { return media_config.enable_dscp; }
|
||||
void set_dscp(bool enable) { media_config.enable_dscp = enable; }
|
||||
|
||||
bool stats_timestamp_with_environment_clock() const {
|
||||
return media_config.stats_timestamp_with_environment_clock;
|
||||
}
|
||||
void set_stats_timestamp_with_environment_clock(bool enable) {
|
||||
media_config.stats_timestamp_with_environment_clock = enable;
|
||||
}
|
||||
|
||||
bool cpu_adaptation() const {
|
||||
return media_config.video.enable_cpu_adaptation;
|
||||
}
|
||||
|
|
@ -842,7 +849,7 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
// TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
|
||||
// is removed; remove default implementation once upstream is updated.
|
||||
virtual RTCError RemoveTrackOrError(
|
||||
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
||||
rtc::scoped_refptr<RtpSenderInterface> /* sender */) {
|
||||
RTC_CHECK_NOTREACHED();
|
||||
return RTCError();
|
||||
}
|
||||
|
|
@ -983,8 +990,8 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
// in SDP, so it should be done before CreateOffer is called, if the
|
||||
// application plans to use data channels.
|
||||
virtual RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
|
||||
CreateDataChannelOrError(const std::string& label,
|
||||
const DataChannelInit* config) {
|
||||
CreateDataChannelOrError(const std::string& /* label */,
|
||||
const DataChannelInit* /* config */) {
|
||||
return RTCError(RTCErrorType::INTERNAL_ERROR, "dummy function called");
|
||||
}
|
||||
// TODO(crbug.com/788659): Remove "virtual" below and default implementation
|
||||
|
|
@ -1053,15 +1060,17 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
// The observer is invoked as soon as the operation completes, which could be
|
||||
// before or after the SetLocalDescription() method has exited.
|
||||
virtual void SetLocalDescription(
|
||||
std::unique_ptr<SessionDescriptionInterface> desc,
|
||||
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
|
||||
std::unique_ptr<SessionDescriptionInterface> /* desc */,
|
||||
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> /* observer */) {
|
||||
}
|
||||
// Creates an offer or answer (depending on current signaling state) and sets
|
||||
// it as the local session description.
|
||||
//
|
||||
// The observer is invoked as soon as the operation completes, which could be
|
||||
// before or after the SetLocalDescription() method has exited.
|
||||
virtual void SetLocalDescription(
|
||||
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
|
||||
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> /* observer */) {
|
||||
}
|
||||
// Like SetLocalDescription() above, but the observer is invoked with a delay
|
||||
// after the operation completes. This helps avoid recursive calls by the
|
||||
// observer but also makes it possible for states to change in-between the
|
||||
|
|
@ -1071,7 +1080,8 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
// ones taking SetLocalDescriptionObserverInterface as argument.
|
||||
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
|
||||
SessionDescriptionInterface* desc) = 0;
|
||||
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer) {}
|
||||
virtual void SetLocalDescription(
|
||||
SetSessionDescriptionObserver* /* observer */) {}
|
||||
|
||||
// Sets the remote session description.
|
||||
//
|
||||
|
|
@ -1090,8 +1100,9 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
// for synchronizing peer connection states to the application.
|
||||
// TODO(https://crbug.com/webrtc/11798): Delete this method in favor of the
|
||||
// ones taking SetRemoteDescriptionObserverInterface as argument.
|
||||
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
|
||||
SessionDescriptionInterface* desc) {}
|
||||
virtual void SetRemoteDescription(
|
||||
SetSessionDescriptionObserver* /* observer */,
|
||||
SessionDescriptionInterface* /* desc */) {}
|
||||
|
||||
// According to spec, we must only fire "negotiationneeded" if the Operations
|
||||
// Chain is empty. This method takes care of validating an event previously
|
||||
|
|
@ -1136,8 +1147,9 @@ class RTC_EXPORT PeerConnectionInterface : public webrtc::RefCountInterface {
|
|||
virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
|
||||
// TODO(hbos): Remove default implementation once implemented by downstream
|
||||
// projects.
|
||||
virtual void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate,
|
||||
std::function<void(RTCError)> callback) {}
|
||||
virtual void AddIceCandidate(
|
||||
std::unique_ptr<IceCandidateInterface> /* candidate */,
|
||||
std::function<void(RTCError)> /* callback */) {}
|
||||
|
||||
// Removes a group of remote candidates from the ICE agent. Needed mainly for
|
||||
// continual gathering, to avoid an ever-growing list of candidates as
|
||||
|
|
@ -1310,11 +1322,12 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
PeerConnectionInterface::SignalingState new_state) = 0;
|
||||
|
||||
// Triggered when media is received on a new stream from remote peer.
|
||||
virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
|
||||
virtual void OnAddStream(
|
||||
rtc::scoped_refptr<MediaStreamInterface> /* stream */) {}
|
||||
|
||||
// Triggered when a remote peer closes a stream.
|
||||
virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
|
||||
}
|
||||
virtual void OnRemoveStream(
|
||||
rtc::scoped_refptr<MediaStreamInterface> /* stream */) {}
|
||||
|
||||
// Triggered when a remote peer opens a data channel.
|
||||
virtual void OnDataChannel(
|
||||
|
|
@ -1332,7 +1345,7 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
// PeerConnection::ShouldFireNegotiationNeededEvent() returns true since it is
|
||||
// possible for the event to become invalidated by operations subsequently
|
||||
// chained.
|
||||
virtual void OnNegotiationNeededEvent(uint32_t event_id) {}
|
||||
virtual void OnNegotiationNeededEvent(uint32_t /* event_id */) {}
|
||||
|
||||
// Called any time the legacy IceConnectionState changes.
|
||||
//
|
||||
|
|
@ -1343,15 +1356,15 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
//
|
||||
// TODO(jonasolsson): deprecate and remove this.
|
||||
virtual void OnIceConnectionChange(
|
||||
PeerConnectionInterface::IceConnectionState new_state) {}
|
||||
PeerConnectionInterface::IceConnectionState /* new_state */) {}
|
||||
|
||||
// Called any time the standards-compliant IceConnectionState changes.
|
||||
virtual void OnStandardizedIceConnectionChange(
|
||||
PeerConnectionInterface::IceConnectionState new_state) {}
|
||||
PeerConnectionInterface::IceConnectionState /* new_state */) {}
|
||||
|
||||
// Called any time the PeerConnectionState changes.
|
||||
virtual void OnConnectionChange(
|
||||
PeerConnectionInterface::PeerConnectionState new_state) {}
|
||||
PeerConnectionInterface::PeerConnectionState /* new_state */) {}
|
||||
|
||||
// Called any time the IceGatheringState changes.
|
||||
virtual void OnIceGatheringChange(
|
||||
|
|
@ -1362,24 +1375,24 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
|
||||
// Gathering of an ICE candidate failed.
|
||||
// See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
|
||||
virtual void OnIceCandidateError(const std::string& address,
|
||||
int port,
|
||||
const std::string& url,
|
||||
int error_code,
|
||||
const std::string& error_text) {}
|
||||
virtual void OnIceCandidateError(const std::string& /* address */,
|
||||
int /* port */,
|
||||
const std::string& /* url */,
|
||||
int /* error_code */,
|
||||
const std::string& /* error_text */) {}
|
||||
|
||||
// Ice candidates have been removed.
|
||||
// TODO(honghaiz): Make this a pure virtual method when all its subclasses
|
||||
// implement it.
|
||||
virtual void OnIceCandidatesRemoved(
|
||||
const std::vector<cricket::Candidate>& candidates) {}
|
||||
const std::vector<cricket::Candidate>& /* candidates */) {}
|
||||
|
||||
// Called when the ICE connection receiving status changes.
|
||||
virtual void OnIceConnectionReceivingChange(bool receiving) {}
|
||||
virtual void OnIceConnectionReceivingChange(bool /* receiving */) {}
|
||||
|
||||
// Called when the selected candidate pair for the ICE connection changes.
|
||||
virtual void OnIceSelectedCandidatePairChanged(
|
||||
const cricket::CandidatePairChangeEvent& event) {}
|
||||
const cricket::CandidatePairChangeEvent& /* event */) {}
|
||||
|
||||
// This is called when a receiver and its track are created.
|
||||
// TODO(zhihuang): Make this pure virtual when all subclasses implement it.
|
||||
|
|
@ -1387,8 +1400,9 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
// Plan users should prefer OnTrack, OnAddTrack is only called as backwards
|
||||
// compatibility (and is called in the exact same situations as OnTrack).
|
||||
virtual void OnAddTrack(
|
||||
rtc::scoped_refptr<RtpReceiverInterface> receiver,
|
||||
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
|
||||
rtc::scoped_refptr<RtpReceiverInterface> /* receiver */,
|
||||
const std::vector<
|
||||
rtc::scoped_refptr<MediaStreamInterface>>& /* streams */) {}
|
||||
|
||||
// This is called when signaling indicates a transceiver will be receiving
|
||||
// media from the remote endpoint. This is fired during a call to
|
||||
|
|
@ -1400,7 +1414,7 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
// RTCSessionDescription" algorithm:
|
||||
// https://w3c.github.io/webrtc-pc/#set-description
|
||||
virtual void OnTrack(
|
||||
rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
|
||||
rtc::scoped_refptr<RtpTransceiverInterface> /* transceiver */) {}
|
||||
|
||||
// Called when signaling indicates that media will no longer be received on a
|
||||
// track.
|
||||
|
|
@ -1411,7 +1425,7 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
// https://w3c.github.io/webrtc-pc/#process-remote-track-removal
|
||||
// TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
|
||||
virtual void OnRemoveTrack(
|
||||
rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
|
||||
rtc::scoped_refptr<RtpReceiverInterface> /* receiver */) {}
|
||||
|
||||
// Called when an interesting usage is detected by WebRTC.
|
||||
// An appropriate action is to add information about the context of the
|
||||
|
|
@ -1419,7 +1433,7 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
|
|||
// log function.
|
||||
// The heuristics for defining what constitutes "interesting" are
|
||||
// implementation-defined.
|
||||
virtual void OnInterestingUsage(int usage_pattern) {}
|
||||
virtual void OnInterestingUsage(int /* usage_pattern */) {}
|
||||
|
||||
// RingRTC change to receive RTP data
|
||||
void OnRtpPacket(const RtpPacketReceived& rtp_packet) override {}
|
||||
|
|
@ -1523,7 +1537,10 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
|
|||
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory;
|
||||
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory;
|
||||
rtc::scoped_refptr<AudioMixer> audio_mixer;
|
||||
// TODO: bugs.webrtc.org/369904700 - Deprecate `audio_processing` in favor
|
||||
// of `audio_processing_builder`.
|
||||
rtc::scoped_refptr<AudioProcessing> audio_processing;
|
||||
std::unique_ptr<AudioProcessingBuilderInterface> audio_processing_builder;
|
||||
std::unique_ptr<AudioFrameProcessor> audio_frame_processor;
|
||||
std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
|
||||
std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
|
||||
|
|
@ -1663,7 +1680,7 @@ class RTC_EXPORT PeerConnectionFactoryInterface
|
|||
// StopAecDump function is called.
|
||||
// TODO(webrtc:6463): Delete default implementation when downstream mocks
|
||||
// classes are updated.
|
||||
virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
|
||||
virtual bool StartAecDump(FILE* /* file */, int64_t /* max_size_bytes */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -60,7 +60,7 @@ class RtcEventLogNull final : public RtcEventLog {
|
|||
bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
|
||||
int64_t output_period_ms) override;
|
||||
void StopLogging() override {}
|
||||
void Log(std::unique_ptr<RtcEvent> event) override {}
|
||||
void Log(std::unique_ptr<RtcEvent> /* event */) override {}
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -27,7 +27,7 @@ class RTC_EXPORT RtcEventLogFactory : public RtcEventLogFactoryInterface {
|
|||
RtcEventLogFactory() = default;
|
||||
|
||||
[[deprecated("Use default constructor")]] //
|
||||
explicit RtcEventLogFactory(TaskQueueFactory* task_queue_factory) {}
|
||||
explicit RtcEventLogFactory(TaskQueueFactory* /* task_queue_factory */) {}
|
||||
|
||||
~RtcEventLogFactory() override = default;
|
||||
|
||||
|
|
|
|||
|
|
@ -32,7 +32,7 @@ class RtpPacketSender {
|
|||
// Clear any pending packets with the given SSRC from the queue.
|
||||
// TODO(crbug.com/1395081): Make pure virtual when downstream code has been
|
||||
// updated.
|
||||
virtual void RemovePacketsForSsrc(uint32_t ssrc) {}
|
||||
virtual void RemovePacketsForSsrc(uint32_t /* ssrc */) {}
|
||||
};
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -76,6 +76,15 @@ RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
|
|||
int preferred_id,
|
||||
RtpTransceiverDirection direction)
|
||||
: uri(uri), preferred_id(preferred_id), direction(direction) {}
|
||||
RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
|
||||
absl::string_view uri,
|
||||
int preferred_id,
|
||||
bool preferred_encrypt,
|
||||
RtpTransceiverDirection direction)
|
||||
: uri(uri),
|
||||
preferred_id(preferred_id),
|
||||
preferred_encrypt(preferred_encrypt),
|
||||
direction(direction) {}
|
||||
RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() = default;
|
||||
|
||||
RtpExtension::RtpExtension() = default;
|
||||
|
|
|
|||
|
|
@ -212,7 +212,6 @@ struct RTC_EXPORT RtpHeaderExtensionCapability {
|
|||
std::optional<int> preferred_id;
|
||||
|
||||
// If true, it's preferred that the value in the header is encrypted.
|
||||
// TODO(deadbeef): Not implemented.
|
||||
bool preferred_encrypt = false;
|
||||
|
||||
// The direction of the extension. The kStopped value is only used with
|
||||
|
|
@ -227,6 +226,10 @@ struct RTC_EXPORT RtpHeaderExtensionCapability {
|
|||
RtpHeaderExtensionCapability(absl::string_view uri,
|
||||
int preferred_id,
|
||||
RtpTransceiverDirection direction);
|
||||
RtpHeaderExtensionCapability(absl::string_view uri,
|
||||
int preferred_id,
|
||||
bool preferred_encrypt,
|
||||
RtpTransceiverDirection direction);
|
||||
~RtpHeaderExtensionCapability();
|
||||
|
||||
bool operator==(const RtpHeaderExtensionCapability& o) const {
|
||||
|
|
@ -494,23 +497,16 @@ struct RTC_EXPORT RtpEncodingParameters {
|
|||
// https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters
|
||||
std::optional<std::string> scalability_mode;
|
||||
|
||||
// Requested encode resolution.
|
||||
// This is an alternative API to `scale_resolution_down_by` but expressed in
|
||||
// absolute terms (max width and max height) as opposed to relative terms (a
|
||||
// scaling factor that is relative to the input frame size).
|
||||
//
|
||||
// This field provides an alternative to `scale_resolution_down_by`
|
||||
// that is not dependent on the video source.
|
||||
// If both `scale_resolution_down_by` and `scale_resolution_down_to` are
|
||||
// specified, the "scale by" value is ignored.
|
||||
//
|
||||
// When setting requested_resolution it is not necessary to adapt the
|
||||
// video source using OnOutputFormatRequest, since the VideoStreamEncoder
|
||||
// will apply downscaling if necessary. requested_resolution will also be
|
||||
// propagated to the video source, this allows downscaling earlier in the
|
||||
// pipeline which can be beneficial if the source is consumed by multiple
|
||||
// encoders, but is not strictly necessary.
|
||||
//
|
||||
// The `requested_resolution` is subject to resource adaptation.
|
||||
//
|
||||
// It is an error to set both `requested_resolution` and
|
||||
// `scale_resolution_down_by`.
|
||||
std::optional<Resolution> requested_resolution;
|
||||
// See spec:
|
||||
// https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-scaleresolutiondownto
|
||||
std::optional<Resolution> scale_resolution_down_to;
|
||||
|
||||
// For an RtpSender, set to true to cause this encoding to be encoded and
|
||||
// sent, and false for it not to be encoded and sent. This allows control
|
||||
|
|
@ -543,7 +539,8 @@ struct RTC_EXPORT RtpEncodingParameters {
|
|||
scale_resolution_down_by == o.scale_resolution_down_by &&
|
||||
active == o.active && rid == o.rid &&
|
||||
adaptive_ptime == o.adaptive_ptime &&
|
||||
requested_resolution == o.requested_resolution && codec == o.codec;
|
||||
scale_resolution_down_to == o.scale_resolution_down_to &&
|
||||
codec == o.codec;
|
||||
}
|
||||
bool operator!=(const RtpEncodingParameters& o) const {
|
||||
return !(*this == o);
|
||||
|
|
|
|||
|
|
@ -36,7 +36,7 @@ std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
|
|||
}
|
||||
|
||||
void RtpReceiverInterface::SetFrameDecryptor(
|
||||
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {}
|
||||
rtc::scoped_refptr<FrameDecryptorInterface> /* frame_decryptor */) {}
|
||||
|
||||
rtc::scoped_refptr<FrameDecryptorInterface>
|
||||
RtpReceiverInterface::GetFrameDecryptor() const {
|
||||
|
|
@ -49,6 +49,6 @@ RtpReceiverInterface::dtls_transport() const {
|
|||
}
|
||||
|
||||
void RtpReceiverInterface::SetFrameTransformer(
|
||||
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {}
|
||||
rtc::scoped_refptr<FrameTransformerInterface> /* frame_transformer */) {}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -80,7 +80,9 @@ class RTC_EXPORT RtpReceiverInterface : public webrtc::RefCountInterface,
|
|||
virtual RtpParameters GetParameters() const = 0;
|
||||
// TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
|
||||
// Currently, doesn't support changing any parameters.
|
||||
virtual bool SetParameters(const RtpParameters& parameters) { return false; }
|
||||
virtual bool SetParameters(const RtpParameters& /* parameters */) {
|
||||
return false;
|
||||
}
|
||||
|
||||
// Does not take ownership of observer.
|
||||
// Must call SetObserver(nullptr) before the observer is destroyed.
|
||||
|
|
|
|||
|
|
@ -15,8 +15,9 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
void RtpSenderInterface::SetParametersAsync(const RtpParameters& parameters,
|
||||
SetParametersCallback callback) {
|
||||
void RtpSenderInterface::SetParametersAsync(
|
||||
const RtpParameters& /* parameters */,
|
||||
SetParametersCallback /* callback */) {
|
||||
RTC_DCHECK_NOTREACHED() << "Default implementation called";
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -36,6 +36,16 @@
|
|||
|
||||
namespace webrtc {
|
||||
|
||||
class RtpSenderObserverInterface {
|
||||
public:
|
||||
// The observer is called when the first media packet is sent for the observed
|
||||
// sender. It is called immediately if the first packet was already sent.
|
||||
virtual void OnFirstPacketSent(cricket::MediaType media_type) = 0;
|
||||
|
||||
protected:
|
||||
virtual ~RtpSenderObserverInterface() {}
|
||||
};
|
||||
|
||||
using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
|
||||
|
||||
class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,
|
||||
|
|
@ -88,6 +98,12 @@ class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,
|
|||
virtual void SetParametersAsync(const RtpParameters& parameters,
|
||||
SetParametersCallback callback);
|
||||
|
||||
// Sets an observer which gets a callback when the first media packet is sent
|
||||
// for this sender.
|
||||
// Does not take ownership of observer.
|
||||
// Must call SetObserver(nullptr) before the observer is destroyed.
|
||||
virtual void SetObserver(RtpSenderObserverInterface* /* observer */) {}
|
||||
|
||||
// Returns null for a video sender.
|
||||
virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
|
||||
|
||||
|
|
@ -119,7 +135,7 @@ class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,
|
|||
// Default implementation of SetFrameTransformer.
|
||||
// TODO: bugs.webrtc.org/15929 - remove when all implementations are good
|
||||
void SetFrameTransformer(rtc::scoped_refptr<FrameTransformerInterface>
|
||||
frame_transformer) override {}
|
||||
/* frame_transformer */) override {}
|
||||
|
||||
protected:
|
||||
~RtpSenderInterface() override = default;
|
||||
|
|
|
|||
|
|
@ -10,6 +10,8 @@
|
|||
|
||||
#include "api/rtp_transceiver_interface.h"
|
||||
|
||||
#include <optional>
|
||||
|
||||
#include "api/rtc_error.h"
|
||||
#include "api/rtp_transceiver_direction.h"
|
||||
#include "rtc_base/checks.h"
|
||||
|
|
@ -54,7 +56,7 @@ void RtpTransceiverInterface::SetDirection(
|
|||
}
|
||||
|
||||
RTCError RtpTransceiverInterface::SetDirectionWithError(
|
||||
RtpTransceiverDirection new_direction) {
|
||||
RtpTransceiverDirection /* new_direction */) {
|
||||
RTC_DCHECK_NOTREACHED() << "Default implementation called";
|
||||
return RTCError::OK();
|
||||
}
|
||||
|
|
|
|||
|
|
@ -22,13 +22,13 @@
|
|||
// };
|
||||
//
|
||||
// void some_function() {
|
||||
// scoped_refptr<MyFoo> foo = new MyFoo();
|
||||
// scoped_refptr<MyFoo> foo = make_ref_counted<MyFoo>();
|
||||
// foo->Method(param);
|
||||
// // `foo` is released when this function returns
|
||||
// }
|
||||
//
|
||||
// void some_other_function() {
|
||||
// scoped_refptr<MyFoo> foo = new MyFoo();
|
||||
// scoped_refptr<MyFoo> foo = make_ref_counted<MyFoo>();
|
||||
// ...
|
||||
// foo = nullptr; // explicitly releases `foo`
|
||||
// ...
|
||||
|
|
@ -41,7 +41,7 @@
|
|||
// references between the two objects, like so:
|
||||
//
|
||||
// {
|
||||
// scoped_refptr<MyFoo> a = new MyFoo();
|
||||
// scoped_refptr<MyFoo> a = make_ref_counted<MyFoo>();
|
||||
// scoped_refptr<MyFoo> b;
|
||||
//
|
||||
// b.swap(a);
|
||||
|
|
@ -52,7 +52,7 @@
|
|||
// object, simply use the assignment operator:
|
||||
//
|
||||
// {
|
||||
// scoped_refptr<MyFoo> a = new MyFoo();
|
||||
// scoped_refptr<MyFoo> a = make_ref_counted<MyFoo>();
|
||||
// scoped_refptr<MyFoo> b;
|
||||
//
|
||||
// b = a;
|
||||
|
|
@ -66,17 +66,20 @@
|
|||
#include <cstddef>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
template <class T>
|
||||
class scoped_refptr {
|
||||
class ABSL_NULLABILITY_COMPATIBLE scoped_refptr {
|
||||
public:
|
||||
typedef T element_type;
|
||||
using absl_nullability_compatible = void;
|
||||
using element_type = T;
|
||||
|
||||
scoped_refptr() : ptr_(nullptr) {}
|
||||
scoped_refptr(std::nullptr_t) : ptr_(nullptr) {} // NOLINT(runtime/explicit)
|
||||
|
||||
explicit scoped_refptr(T* p) : ptr_(p) {
|
||||
explicit scoped_refptr(absl::Nullable<T*> p) : ptr_(p) {
|
||||
if (ptr_)
|
||||
ptr_->AddRef();
|
||||
}
|
||||
|
|
@ -119,7 +122,7 @@ class scoped_refptr {
|
|||
return retVal;
|
||||
}
|
||||
|
||||
scoped_refptr<T>& operator=(T* p) {
|
||||
scoped_refptr<T>& operator=(absl::Nullable<T*> p) {
|
||||
// AddRef first so that self assignment should work
|
||||
if (p)
|
||||
p->AddRef();
|
||||
|
|
@ -149,7 +152,7 @@ class scoped_refptr {
|
|||
return *this;
|
||||
}
|
||||
|
||||
void swap(T** pp) noexcept {
|
||||
void swap(absl::Nonnull<T**> pp) noexcept {
|
||||
T* p = ptr_;
|
||||
ptr_ = *pp;
|
||||
*pp = p;
|
||||
|
|
@ -219,9 +222,8 @@ bool operator<(const scoped_refptr<T>& a, const scoped_refptr<U>& b) {
|
|||
|
||||
namespace rtc {
|
||||
// Backwards compatible alias.
|
||||
// TODO(bugs.webrtc.org/15622): Deprecate and remove.
|
||||
template <typename T>
|
||||
using scoped_refptr = webrtc::scoped_refptr<T>;
|
||||
// TODO: bugs.webrtc.org/42225969 - Deprecate and remove.
|
||||
using ::webrtc::scoped_refptr;
|
||||
} // namespace rtc
|
||||
|
||||
#endif // API_SCOPED_REFPTR_H_
|
||||
|
|
|
|||
|
|
@ -10,6 +10,7 @@
|
|||
|
||||
#include "api/sctp_transport_interface.h"
|
||||
|
||||
#include <optional>
|
||||
#include <utility>
|
||||
|
||||
#include "api/dtls_transport_interface.h"
|
||||
|
|
|
|||
|
|
@ -297,6 +297,9 @@ class RTC_EXPORT RTCInboundRtpStreamStats final
|
|||
std::optional<uint32_t> pli_count;
|
||||
std::optional<uint32_t> nack_count;
|
||||
std::optional<uint64_t> qp_sum;
|
||||
std::optional<double> total_corruption_probability;
|
||||
std::optional<double> total_squared_corruption_probability;
|
||||
std::optional<uint64_t> corruption_measurements;
|
||||
// This is a remnant of the legacy getStats() API. When the "video-timing"
|
||||
// header extension is used,
|
||||
// https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/,
|
||||
|
|
|
|||
|
|
@ -16,7 +16,7 @@
|
|||
namespace webrtc {
|
||||
|
||||
std::unique_ptr<TaskQueueFactory> CreateDefaultTaskQueueFactory(
|
||||
const FieldTrialsView* field_trials) {
|
||||
const FieldTrialsView* /* field_trials */) {
|
||||
return CreateTaskQueueGcdFactory();
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -13,26 +13,14 @@
|
|||
#include "api/field_trials_view.h"
|
||||
#include "api/task_queue/task_queue_factory.h"
|
||||
#include "api/transport/field_trial_based_config.h"
|
||||
#include "rtc_base/logging.h"
|
||||
#include "rtc_base/memory/always_valid_pointer.h"
|
||||
#include "rtc_base/task_queue_libevent.h"
|
||||
#include "rtc_base/task_queue_stdlib.h"
|
||||
|
||||
namespace webrtc {
|
||||
|
||||
std::unique_ptr<TaskQueueFactory> CreateDefaultTaskQueueFactory(
|
||||
const FieldTrialsView* field_trials_view) {
|
||||
AlwaysValidPointer<const FieldTrialsView, FieldTrialBasedConfig> field_trials(
|
||||
field_trials_view);
|
||||
if (field_trials->IsEnabled("WebRTC-TaskQueue-ReplaceLibeventWithStdlib")) {
|
||||
RTC_LOG(LS_INFO) << "WebRTC-TaskQueue-ReplaceLibeventWithStdlib: "
|
||||
<< "using TaskQueueStdlibFactory.";
|
||||
return CreateTaskQueueStdlibFactory();
|
||||
}
|
||||
|
||||
RTC_LOG(LS_INFO) << "WebRTC-TaskQueue-ReplaceLibeventWithStdlib: "
|
||||
<< "using TaskQueueLibeventFactory.";
|
||||
return CreateTaskQueueLibeventFactory();
|
||||
return CreateTaskQueueStdlibFactory();
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -13,25 +13,31 @@
|
|||
#include <memory>
|
||||
#include <utility>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
#include "modules/audio_processing/test/audioproc_float_impl.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
||||
int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
|
||||
int argc,
|
||||
char* argv[]) {
|
||||
return AudioprocFloatImpl(std::move(audio_processing), argc, argv);
|
||||
int AudioprocFloat(int argc, char* argv[]) {
|
||||
return AudioprocFloatImpl(std::make_unique<BuiltinAudioProcessingBuilder>(),
|
||||
argc, argv);
|
||||
}
|
||||
|
||||
int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
|
||||
int argc,
|
||||
char* argv[]) {
|
||||
return AudioprocFloatImpl(std::move(ap_builder), argc, argv,
|
||||
/*input_aecdump=*/"",
|
||||
/*processed_capture_samples=*/nullptr);
|
||||
int AudioprocFloat(
|
||||
absl::Nonnull<std::unique_ptr<BuiltinAudioProcessingBuilder>> ap_builder,
|
||||
int argc,
|
||||
char* argv[]) {
|
||||
return AudioprocFloatImpl(std::move(ap_builder), argc, argv);
|
||||
}
|
||||
|
||||
int AudioprocFloat(
|
||||
absl::Nonnull<std::unique_ptr<AudioProcessingBuilderInterface>> ap_builder,
|
||||
int argc,
|
||||
char* argv[]) {
|
||||
return AudioprocFloatImpl(std::move(ap_builder), argc, argv);
|
||||
}
|
||||
|
||||
} // namespace test
|
||||
|
|
|
|||
|
|
@ -13,8 +13,9 @@
|
|||
|
||||
#include <memory>
|
||||
|
||||
#include "absl/base/nullability.h"
|
||||
#include "api/audio/audio_processing.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/audio/builtin_audio_processing_builder.h"
|
||||
|
||||
namespace webrtc {
|
||||
namespace test {
|
||||
|
|
@ -22,36 +23,26 @@ namespace test {
|
|||
// This is an interface for the audio processing simulation utility. This
|
||||
// utility can be used to simulate the audioprocessing module using a recording
|
||||
// (either an AEC dump or wav files), and generate the output as a wav file.
|
||||
// Any audio_processing object specified in the input is used for the
|
||||
// simulation. The optional `audio_processing` object provides the
|
||||
// AudioProcessing instance that is used during the simulation. Note that when
|
||||
// the audio_processing object is specified all functionality that relies on
|
||||
// using the AudioProcessingBuilder is deactivated, since the AudioProcessing
|
||||
// object is already created and the builder is not used in the simulation. It
|
||||
// is needed to pass the command line flags as `argc` and `argv`, so these can
|
||||
// be interpreted properly by the utility. To see a list of all supported
|
||||
// command line flags, run the executable with the '--help' flag.
|
||||
int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
|
||||
int argc,
|
||||
char* argv[]);
|
||||
|
||||
// This is an interface for the audio processing simulation utility. This
|
||||
// utility can be used to simulate the audioprocessing module using a recording
|
||||
// (either an AEC dump or wav files), and generate the output as a wav file.
|
||||
// The `ap_builder` object will be used to create the AudioProcessing instance
|
||||
// that is used during the simulation. The `ap_builder` supports setting of
|
||||
// injectable components, which will be passed on to the created AudioProcessing
|
||||
// instance. It is needed to pass the command line flags as `argc` and `argv`,
|
||||
// so these can be interpreted properly by the utility.
|
||||
// To get a fully-working audioproc_f utility, all that is needed is to write a
|
||||
// main function, create an AudioProcessingBuilder, optionally set custom
|
||||
// processing components on it, and pass the builder together with the command
|
||||
// line arguments into this function.
|
||||
// To see a list of all supported command line flags, run the executable with
|
||||
// the '--help' flag.
|
||||
int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
|
||||
int argc,
|
||||
char* argv[]);
|
||||
//
|
||||
// It is needed to pass the command line flags as `argc` and `argv`, so these
|
||||
// can be interpreted properly by the utility. To see a list of all supported
|
||||
// command line flags, run the executable with the '--helpfull' flag.
|
||||
//
|
||||
// The optional `ap_builder` object will be used to create the AudioProcessing
|
||||
// instance that is used during the simulation. BuiltinAudioProcessingBuilder
|
||||
// `ap_builder` supports setting of injectable components, which will be passed
|
||||
// on to the created AudioProcessing instance. When generic
|
||||
// `AudioProcessingBuilderInterface` is used, all functionality that relies on
|
||||
// using the BuiltinAudioProcessingBuilder is deactivated.
|
||||
int AudioprocFloat(int argc, char* argv[]);
|
||||
int AudioprocFloat(
|
||||
absl::Nonnull<std::unique_ptr<BuiltinAudioProcessingBuilder>> ap_builder,
|
||||
int argc,
|
||||
char* argv[]);
|
||||
int AudioprocFloat(
|
||||
absl::Nonnull<std::unique_ptr<AudioProcessingBuilderInterface>> ap_builder,
|
||||
int argc,
|
||||
char* argv[]);
|
||||
|
||||
} // namespace test
|
||||
} // namespace webrtc
|
||||
|
|
|
|||
|
|
@ -25,9 +25,9 @@ FakeFrameDecryptor::FakeFrameDecryptor(uint8_t fake_key,
|
|||
: fake_key_(fake_key), expected_postfix_byte_(expected_postfix_byte) {}
|
||||
|
||||
FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt(
|
||||
cricket::MediaType media_type,
|
||||
const std::vector<uint32_t>& csrcs,
|
||||
rtc::ArrayView<const uint8_t> additional_data,
|
||||
cricket::MediaType /* media_type */,
|
||||
const std::vector<uint32_t>& /* csrcs */,
|
||||
rtc::ArrayView<const uint8_t> /* additional_data */,
|
||||
rtc::ArrayView<const uint8_t> encrypted_frame,
|
||||
rtc::ArrayView<uint8_t> frame) {
|
||||
if (fail_decryption_) {
|
||||
|
|
@ -47,7 +47,7 @@ FakeFrameDecryptor::Result FakeFrameDecryptor::Decrypt(
|
|||
}
|
||||
|
||||
size_t FakeFrameDecryptor::GetMaxPlaintextByteSize(
|
||||
cricket::MediaType media_type,
|
||||
cricket::MediaType /* media_type */,
|
||||
size_t encrypted_frame_size) {
|
||||
return encrypted_frame_size - 1;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -22,12 +22,13 @@ FakeFrameEncryptor::FakeFrameEncryptor(uint8_t fake_key, uint8_t postfix_byte)
|
|||
: fake_key_(fake_key), postfix_byte_(postfix_byte) {}
|
||||
|
||||
// FrameEncryptorInterface implementation
|
||||
int FakeFrameEncryptor::Encrypt(cricket::MediaType media_type,
|
||||
uint32_t ssrc,
|
||||
rtc::ArrayView<const uint8_t> additional_data,
|
||||
rtc::ArrayView<const uint8_t> frame,
|
||||
rtc::ArrayView<uint8_t> encrypted_frame,
|
||||
size_t* bytes_written) {
|
||||
int FakeFrameEncryptor::Encrypt(
|
||||
cricket::MediaType /* media_type */,
|
||||
uint32_t /* ssrc */,
|
||||
rtc::ArrayView<const uint8_t> /* additional_data */,
|
||||
rtc::ArrayView<const uint8_t> frame,
|
||||
rtc::ArrayView<uint8_t> encrypted_frame,
|
||||
size_t* bytes_written) {
|
||||
if (fail_encryption_) {
|
||||
return static_cast<int>(FakeEncryptionStatus::FORCED_FAILURE);
|
||||
}
|
||||
|
|
@ -43,7 +44,7 @@ int FakeFrameEncryptor::Encrypt(cricket::MediaType media_type,
|
|||
}
|
||||
|
||||
size_t FakeFrameEncryptor::GetMaxCiphertextByteSize(
|
||||
cricket::MediaType media_type,
|
||||
cricket::MediaType /* media_type */,
|
||||
size_t frame_size) {
|
||||
return frame_size + 1;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -45,9 +45,9 @@ class MockDtmfSender : public DtmfSenderInterface {
|
|||
(override));
|
||||
MOCK_METHOD(void, UnregisterObserver, (), (override));
|
||||
MOCK_METHOD(bool, CanInsertDtmf, (), (override));
|
||||
MOCK_METHOD(std::string, tones, (), (const override));
|
||||
MOCK_METHOD(int, duration, (), (const override));
|
||||
MOCK_METHOD(int, inter_tone_gap, (), (const override));
|
||||
MOCK_METHOD(std::string, tones, (), (const, override));
|
||||
MOCK_METHOD(int, duration, (), (const, override));
|
||||
MOCK_METHOD(int, inter_tone_gap, (), (const, override));
|
||||
|
||||
protected:
|
||||
MockDtmfSender() = default;
|
||||
|
|
|
|||
|
|
@ -89,7 +89,7 @@ class MockAudioTrack : public rtc::RefCountedObject<AudioTrackInterface> {
|
|||
|
||||
class MockMediaStream : public MediaStreamInterface {
|
||||
public:
|
||||
MOCK_METHOD(std::string, id, (), (const override));
|
||||
MOCK_METHOD(std::string, id, (), (const, override));
|
||||
MOCK_METHOD(AudioTrackVector, GetAudioTracks, (), (override));
|
||||
MOCK_METHOD(VideoTrackVector, GetVideoTracks, (), (override));
|
||||
MOCK_METHOD(rtc::scoped_refptr<AudioTrackInterface>,
|
||||
|
|
@ -116,11 +116,6 @@ class MockMediaStream : public MediaStreamInterface {
|
|||
RemoveTrack,
|
||||
(rtc::scoped_refptr<VideoTrackInterface> track),
|
||||
(override));
|
||||
// Old AddTrack/RemoveTrack methods - slated for removal
|
||||
MOCK_METHOD(bool, AddTrack, (AudioTrackInterface * track), (override));
|
||||
MOCK_METHOD(bool, AddTrack, (VideoTrackInterface * track), (override));
|
||||
MOCK_METHOD(bool, RemoveTrack, (AudioTrackInterface * track), (override));
|
||||
MOCK_METHOD(bool, RemoveTrack, (VideoTrackInterface * track), (override));
|
||||
MOCK_METHOD(void,
|
||||
RegisterObserver,
|
||||
(ObserverInterface * observer),
|
||||
|
|
|
|||
|
|
@ -234,7 +234,7 @@ class MockPeerConnectionInterface : public webrtc::PeerConnectionInterface {
|
|||
(override));
|
||||
MOCK_METHOD(void, StopRtcEventLog, (), (override));
|
||||
MOCK_METHOD(void, Close, (), (override));
|
||||
MOCK_METHOD(rtc::Thread*, signaling_thread, (), (const override));
|
||||
MOCK_METHOD(rtc::Thread*, signaling_thread, (), (const, override));
|
||||
MOCK_METHOD(NetworkControllerInterface*,
|
||||
GetNetworkController,
|
||||
(),
|
||||
|
|
|
|||
|
|
@ -45,10 +45,8 @@ class MockRtpSender : public RtpSenderInterface {
|
|||
track,
|
||||
(),
|
||||
(const, override));
|
||||
MOCK_METHOD(rtc::scoped_refptr<DtlsTransportInterface>,
|
||||
dtls_transport,
|
||||
(),
|
||||
(const override));
|
||||
MOCK_METHOD(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport, (),
|
||||
(const, override));
|
||||
MOCK_METHOD(uint32_t, ssrc, (), (const, override));
|
||||
MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
|
||||
MOCK_METHOD(std::string, id, (), (const, override));
|
||||
|
|
@ -84,6 +82,7 @@ class MockRtpSender : public RtpSenderInterface {
|
|||
SetEncoderSelector,
|
||||
(std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>),
|
||||
(override));
|
||||
MOCK_METHOD(void, SetObserver, (RtpSenderObserverInterface*), (override));
|
||||
};
|
||||
|
||||
static_assert(!std::is_abstract_v<rtc::RefCountedObject<MockRtpSender>>, "");
|
||||
|
|
|
|||
|
|
@ -33,10 +33,8 @@ class MockTransformableAudioFrame : public TransformableAudioFrameInterface {
|
|||
MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
|
||||
MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
|
||||
MOCK_METHOD(std::string, GetMimeType, (), (const, override));
|
||||
MOCK_METHOD(rtc::ArrayView<const uint32_t>,
|
||||
GetContributingSources,
|
||||
(),
|
||||
(const override));
|
||||
MOCK_METHOD(rtc::ArrayView<const uint32_t>, GetContributingSources, (),
|
||||
(const, override));
|
||||
MOCK_METHOD(const std::optional<uint16_t>,
|
||||
SequenceNumber,
|
||||
(),
|
||||
|
|
|
|||
|
|
@ -35,7 +35,7 @@ class MockTransformableFrame : public TransformableFrameInterface {
|
|||
MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
|
||||
MOCK_METHOD(void, SetRTPTimestamp, (uint32_t), (override));
|
||||
MOCK_METHOD(std::optional<webrtc::Timestamp>,
|
||||
GetCaptureTimeIdentifier,
|
||||
GetPresentationTimestamp,
|
||||
(),
|
||||
(const, override));
|
||||
MOCK_METHOD(std::string, GetMimeType, (), (const, override));
|
||||
|
|
|
|||
|
|
@ -46,7 +46,7 @@ class MockTransformableVideoFrame
|
|||
MOCK_METHOD(std::string, GetMimeType, (), (const, override));
|
||||
MOCK_METHOD(VideoFrameMetadata, Metadata, (), (const, override));
|
||||
MOCK_METHOD(std::optional<Timestamp>,
|
||||
GetCaptureTimeIdentifier,
|
||||
GetPresentationTimestamp,
|
||||
(),
|
||||
(const, override));
|
||||
};
|
||||
|
|
|
|||
|
|
@ -128,10 +128,15 @@ struct Params {
|
|||
std::optional<std::string> name;
|
||||
// If `audio_config` is set audio stream will be configured
|
||||
std::optional<AudioConfig> audio_config;
|
||||
// Flags to set on `cricket::PortAllocator`. These flags will be added
|
||||
// to the default ones that are presented on the port allocator.
|
||||
// Flags to set on `cricket::PortAllocator`. If not set,
|
||||
// cricket::kDefaultPortAllocatorFlags will be used and
|
||||
// cricket::PORTALLOCATOR_DISABLE_TCP will be disabled.
|
||||
//
|
||||
// IMPORTANT: if you use WebRTC Network Emulation
|
||||
// (api/test/network_emulation_manager.h) and set this field, remember to set
|
||||
// cricket::PORTALLOCATOR_DISABLE_TCP to 0.
|
||||
// RingRTC change to avoid incorrect use of flag in non-test code.
|
||||
uint32_t port_allocator_extra_flags = 0;
|
||||
std::optional<uint32_t> port_allocator_flags = 0;
|
||||
// If `rtc_event_log_path` is set, an RTCEventLog will be saved in that
|
||||
// location and it will be available for further analysis.
|
||||
std::optional<std::string> rtc_event_log_path;
|
||||
|
|
|
|||
|
|
@ -240,9 +240,17 @@ PeerConfigurer* PeerConfigurer::SetFieldTrials(
|
|||
|
||||
PeerConfigurer* PeerConfigurer::SetPortAllocatorExtraFlags(
|
||||
uint32_t extra_flags) {
|
||||
params_->port_allocator_extra_flags = extra_flags;
|
||||
params_->port_allocator_flags = cricket::kDefaultPortAllocatorFlags |
|
||||
cricket::PORTALLOCATOR_DISABLE_TCP |
|
||||
extra_flags;
|
||||
return this;
|
||||
}
|
||||
|
||||
PeerConfigurer* PeerConfigurer::SetPortAllocatorFlags(uint32_t flags) {
|
||||
params_->port_allocator_flags = flags;
|
||||
return this;
|
||||
}
|
||||
|
||||
std::unique_ptr<InjectableComponents> PeerConfigurer::ReleaseComponents() {
|
||||
RTC_CHECK(components_);
|
||||
auto components = std::move(components_);
|
||||
|
|
|
|||
|
|
@ -99,9 +99,19 @@ class PeerConfigurer {
|
|||
PeerConfigurer* SetIceTransportFactory(
|
||||
std::unique_ptr<IceTransportFactory> factory);
|
||||
// Flags to set on `cricket::PortAllocator`. These flags will be added
|
||||
// to the default ones that are presented on the port allocator.
|
||||
// For possible values check p2p/base/port_allocator.h.
|
||||
// to the cricket::kDefaultPortAllocatorFlags with
|
||||
// cricket::PORTALLOCATOR_DISABLE_TCP disabled. For possible values check
|
||||
// p2p/base/port_allocator.h.
|
||||
PeerConfigurer* SetPortAllocatorExtraFlags(uint32_t extra_flags);
|
||||
// Flags to set on `cricket::PortAllocator`. These flags will override
|
||||
// the default ones that are presented on the port allocator.
|
||||
//
|
||||
// For possible values check p2p/base/port_allocator.h.
|
||||
//
|
||||
// IMPORTANT: if you use WebRTC Network Emulation
|
||||
// (api/test/network_emulation_manager.h) and set this field, remember to set
|
||||
// cricket::PORTALLOCATOR_DISABLE_TCP to 0.
|
||||
PeerConfigurer* SetPortAllocatorFlags(uint32_t flags);
|
||||
|
||||
// Add new video stream to the call that will be sent from this peer.
|
||||
// Default implementation of video frames generator will be used.
|
||||
|
|
|
|||
|
|
@ -85,7 +85,7 @@ struct TestVideoFrameWriter : public test::VideoFrameWriter {
|
|||
const VideoResolution& resolution)
|
||||
: file_name_prefix(file_name_prefix), resolution(resolution) {}
|
||||
|
||||
bool WriteFrame(const VideoFrame& frame) override { return true; }
|
||||
bool WriteFrame(const VideoFrame& /* frame */) override { return true; }
|
||||
|
||||
void Close() override {}
|
||||
|
||||
|
|
|
|||
Some files were not shown because too many files have changed in this diff Show more
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Reference in a new issue