Revert of Enabling gn check on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
Reason for revert:
ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
#include "webrtc/base/checks.h"
^-------------------
The target:
//webrtc/test:fileutils_unittests
is including a file from the target:
//webrtc/base:rtc_base_approved
It's usually best to depend directly on the destination target.
In some cases, the destination target is considered a subcomponent
of an intermediate target. In this case, the intermediate target
should depend publicly on the destination to forward the ability
to include headers.
Dependency chain (there may also be others):
//webrtc/test:fileutils_unittests -->
//webrtc/test:fileutils --[private]-->
//webrtc/base:rtc_base_approved
Original issue's description:
> Enabling `gn check` on webrtc/test
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2911203002
> Cr-Commit-Position: refs/heads/master@{#18372}
> Committed: db5bb404b0
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2920763002
Cr-Commit-Position: refs/heads/master@{#18375}
This commit is contained in:
parent
11c7b703d7
commit
1a6f143d07
1
.gn
1
.gn
@ -37,7 +37,6 @@ check_targets = [
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"//webrtc/sdk/*",
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"//webrtc/stats/*",
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"//webrtc/system_wrappers/*",
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"//webrtc/test/*",
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"//webrtc/tools/*",
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"//webrtc/video/*",
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"//webrtc/voice_engine/*",
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@ -1103,118 +1103,6 @@ rtc_source_set("neteq_tools_minimal") {
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]
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}
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rtc_source_set("neteq_test_tools") {
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testonly = true
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sources = [
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"neteq/tools/audio_checksum.h",
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"neteq/tools/audio_loop.cc",
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"neteq/tools/audio_loop.h",
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"neteq/tools/constant_pcm_packet_source.cc",
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"neteq/tools/constant_pcm_packet_source.h",
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"neteq/tools/output_audio_file.h",
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"neteq/tools/output_wav_file.h",
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"neteq/tools/rtp_file_source.cc",
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"neteq/tools/rtp_file_source.h",
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"neteq/tools/rtp_generator.cc",
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"neteq/tools/rtp_generator.h",
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]
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public_configs = [ ":neteq_tools_config" ]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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":pcm16b",
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"..:module_api",
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"../..:webrtc_common",
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"../../base:rtc_base_approved",
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"../../base:rtc_base_tests_utils",
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"../../common_audio",
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"../../test:rtp_test_utils",
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"../rtp_rtcp",
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]
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public_deps = [
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":neteq_tools",
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":neteq_tools_minimal",
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]
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if (rtc_enable_protobuf) {
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sources += [
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"neteq/tools/neteq_packet_source_input.cc",
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"neteq/tools/neteq_packet_source_input.h",
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]
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deps += [ ":rtc_event_log_source" ]
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}
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}
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config("neteq_tools_config") {
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include_dirs = [ "tools" ]
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}
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rtc_source_set("neteq_tools") {
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sources = [
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"neteq/tools/fake_decode_from_file.cc",
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"neteq/tools/fake_decode_from_file.h",
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"neteq/tools/input_audio_file.cc",
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"neteq/tools/input_audio_file.h",
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"neteq/tools/neteq_replacement_input.cc",
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"neteq/tools/neteq_replacement_input.h",
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"neteq/tools/resample_input_audio_file.cc",
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"neteq/tools/resample_input_audio_file.h",
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]
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public_configs = [ ":neteq_tools_config" ]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../..:webrtc_common",
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"../../api/audio_codecs:audio_codecs_api",
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"../../base:rtc_base_approved",
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"../../common_audio",
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"../rtp_rtcp",
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]
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public_deps = [
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":neteq_tools_minimal",
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]
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}
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if (rtc_enable_protobuf) {
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rtc_static_library("rtc_event_log_source") {
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testonly = true
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# TODO(kjellander): Remove (bugs.webrtc.org/6828)
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# Needs call.h to be moved to webrtc/api first.
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check_includes = false
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sources = [
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"neteq/tools/rtc_event_log_source.cc",
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"neteq/tools/rtc_event_log_source.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../../base:rtc_base_approved",
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"../../logging:rtc_event_log_parser",
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]
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public_deps = [
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"../../logging:rtc_event_log_proto",
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]
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}
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}
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if (rtc_include_tests) {
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group("audio_coding_tests") {
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testonly = true
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@ -1507,6 +1395,32 @@ if (rtc_include_tests) {
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proto_out_dir = "webrtc/modules/audio_coding/neteq"
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}
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rtc_static_library("rtc_event_log_source") {
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testonly = true
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# TODO(kjellander): Remove (bugs.webrtc.org/6828)
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# Needs call.h to be moved to webrtc/api first.
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check_includes = false
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sources = [
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"neteq/tools/rtc_event_log_source.cc",
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"neteq/tools/rtc_event_log_source.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../../base:rtc_base_approved",
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"../../logging:rtc_event_log_parser",
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]
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public_deps = [
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"../../logging:rtc_event_log_proto",
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]
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}
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rtc_test("neteq_rtpplay") {
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testonly = true
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defines = []
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@ -1628,6 +1542,90 @@ if (rtc_include_tests) {
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]
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}
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config("neteq_tools_config") {
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include_dirs = [ "tools" ]
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}
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rtc_source_set("neteq_tools") {
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sources = [
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"neteq/tools/fake_decode_from_file.cc",
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"neteq/tools/fake_decode_from_file.h",
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"neteq/tools/input_audio_file.cc",
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"neteq/tools/input_audio_file.h",
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"neteq/tools/neteq_replacement_input.cc",
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"neteq/tools/neteq_replacement_input.h",
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"neteq/tools/resample_input_audio_file.cc",
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"neteq/tools/resample_input_audio_file.h",
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]
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public_configs = [ ":neteq_tools_config" ]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"../..:webrtc_common",
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"../../api/audio_codecs:audio_codecs_api",
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"../../base:rtc_base_approved",
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"../../common_audio",
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"../rtp_rtcp",
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]
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public_deps = [
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":neteq_tools_minimal",
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]
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}
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rtc_source_set("neteq_test_tools") {
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testonly = true
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sources = [
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"neteq/tools/audio_checksum.h",
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"neteq/tools/audio_loop.cc",
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"neteq/tools/audio_loop.h",
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"neteq/tools/constant_pcm_packet_source.cc",
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"neteq/tools/constant_pcm_packet_source.h",
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"neteq/tools/output_audio_file.h",
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"neteq/tools/output_wav_file.h",
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"neteq/tools/rtp_file_source.cc",
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"neteq/tools/rtp_file_source.h",
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"neteq/tools/rtp_generator.cc",
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"neteq/tools/rtp_generator.h",
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]
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public_configs = [ ":neteq_tools_config" ]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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":pcm16b",
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"..:module_api",
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"../..:webrtc_common",
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"../../base:rtc_base_approved",
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"../../base:rtc_base_tests_utils",
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"../../common_audio",
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"../../test:rtp_test_utils",
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"../rtp_rtcp",
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]
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public_deps = [
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":neteq_tools",
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":neteq_tools_minimal",
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]
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if (rtc_enable_protobuf) {
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sources += [
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"neteq/tools/neteq_packet_source_input.cc",
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"neteq/tools/neteq_packet_source_input.h",
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]
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deps += [ ":rtc_event_log_source" ]
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}
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}
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rtc_source_set("neteq_test_tools_deprecated") {
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testonly = true
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sources = [
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@ -254,19 +254,6 @@ config("mock_audio_device_config") {
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}
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}
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rtc_source_set("mock_audio_device") {
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testonly = true
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sources = [
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"include/mock_audio_device.h",
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"include/mock_audio_transport.h",
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]
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deps = [
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":audio_device",
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"../../test:test_support",
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]
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all_dependent_configs = [ ":mock_audio_device_config" ]
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}
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if (rtc_include_tests) {
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rtc_source_set("audio_device_unittests") {
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testonly = true
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@ -322,6 +309,19 @@ if (rtc_include_tests) {
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}
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}
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rtc_source_set("mock_audio_device") {
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testonly = true
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sources = [
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"include/mock_audio_device.h",
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"include/mock_audio_transport.h",
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]
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deps = [
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":audio_device",
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"../../test:test_support",
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]
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all_dependent_configs = [ ":mock_audio_device_config" ]
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}
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if (!is_ios) {
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# These tests do not work on ios, see
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# https://bugs.chromium.org/p/webrtc/issues/detail?id=4755
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@ -211,21 +211,6 @@ rtc_source_set("fec_test_helper") {
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}
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}
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rtc_source_set("mock_rtp_rtcp") {
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testonly = true
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sources = [
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"mocks/mock_recovered_packet_receiver.h",
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"mocks/mock_rtcp_rtt_stats.h",
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"mocks/mock_rtp_rtcp.h",
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]
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deps = [
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":rtp_rtcp",
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"..:module_api",
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"../../base:rtc_base_approved",
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"../../test:test_support",
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]
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}
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if (rtc_include_tests) {
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rtc_executable("test_packet_masks_metrics") {
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testonly = true
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@ -265,6 +250,21 @@ if (rtc_include_tests) {
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}
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}
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rtc_source_set("mock_rtp_rtcp") {
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testonly = true
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sources = [
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"mocks/mock_recovered_packet_receiver.h",
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"mocks/mock_rtcp_rtt_stats.h",
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"mocks/mock_rtp_rtcp.h",
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]
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deps = [
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":rtp_rtcp",
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"..:module_api",
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"../../base:rtc_base_approved",
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"../../test:test_support",
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]
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}
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rtc_source_set("rtp_rtcp_unittests") {
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testonly = true
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@ -58,14 +58,9 @@ rtc_source_set("video_test_common") {
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}
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deps = [
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"..:video_stream_api",
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../base:rtc_task_queue",
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"../common_video",
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"../media:rtc_media_base",
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"../modules/video_capture:video_capture_module",
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"../system_wrappers",
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]
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}
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@ -87,7 +82,6 @@ rtc_source_set("rtp_test_utils") {
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deps = [
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../modules/rtp_rtcp",
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"//testing/gtest",
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]
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@ -171,7 +165,6 @@ if (!build_with_chromium) {
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]
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deps = [
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":field_trial",
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"../base:rtc_base_approved",
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"../system_wrappers:metrics_default",
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"//testing/gmock",
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"//testing/gtest",
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@ -195,9 +188,6 @@ if (!build_with_chromium) {
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]
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deps = [
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":test_support",
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":video_test_common",
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../common_video",
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"../system_wrappers",
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@ -262,16 +252,7 @@ if (!build_with_chromium) {
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}
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|
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rtc_test("test_support_unittests") {
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deps = [
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":fake_audio_device",
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":rtp_test_utils",
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"../api:video_frame_api",
|
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"../base:rtc_base_approved",
|
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"../call:call_interfaces",
|
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"../common_audio",
|
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"../modules/rtp_rtcp",
|
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"../system_wrappers",
|
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]
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deps = []
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sources = [
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"fake_audio_device_unittest.cc",
|
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"fake_network_pipe_unittest.cc",
|
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@ -333,16 +314,11 @@ rtc_source_set("fileutils") {
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"testsupport/fileutils.cc",
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"testsupport/fileutils.h",
|
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]
|
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deps = [
|
||||
"..:webrtc_common",
|
||||
"../base:rtc_base_approved",
|
||||
]
|
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if (is_ios) {
|
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sources += [ "testsupport/iosfileutils.mm" ]
|
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deps += [ "../sdk:objc_common" ]
|
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}
|
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if (is_win) {
|
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deps += [ "../base:rtc_base" ]
|
||||
deps = [
|
||||
"../sdk:objc_common",
|
||||
]
|
||||
}
|
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visibility = [ ":*" ]
|
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}
|
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@ -367,7 +343,6 @@ rtc_source_set("fileutils_unittests") {
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]
|
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deps = [
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":fileutils",
|
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":test_support",
|
||||
"//testing/gmock",
|
||||
"//testing/gtest",
|
||||
]
|
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@ -386,12 +361,9 @@ rtc_source_set("direct_transport") {
|
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
||||
}
|
||||
deps = [
|
||||
"..:webrtc_common",
|
||||
"../api:transport_api",
|
||||
"../base:rtc_base_approved",
|
||||
"../call",
|
||||
"../modules/rtp_rtcp",
|
||||
"../system_wrappers",
|
||||
]
|
||||
}
|
||||
|
||||
@ -406,11 +378,8 @@ rtc_source_set("fake_audio_device") {
|
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
|
||||
}
|
||||
deps = [
|
||||
"..:webrtc_common",
|
||||
"../base:rtc_base_approved",
|
||||
"../common_audio:common_audio",
|
||||
"../modules/audio_device:audio_device",
|
||||
"../system_wrappers:system_wrappers",
|
||||
]
|
||||
}
|
||||
|
||||
@ -462,30 +431,15 @@ rtc_source_set("test_common") {
|
||||
":rtp_test_utils",
|
||||
":test_support",
|
||||
":video_test_common",
|
||||
"..:video_stream_api",
|
||||
"..:webrtc_common",
|
||||
"../api:transport_api",
|
||||
"../api:video_frame_api",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../api/audio_codecs:builtin_audio_encoder_factory",
|
||||
"../api/video_codecs:video_codecs_api",
|
||||
"../audio",
|
||||
"../base:rtc_base_approved",
|
||||
"../base:rtc_task_queue",
|
||||
"../call",
|
||||
"../common_video",
|
||||
"../logging:rtc_event_log_api",
|
||||
"../modules/audio_device:mock_audio_device",
|
||||
"../modules/audio_mixer:audio_mixer_impl",
|
||||
"../modules/audio_processing",
|
||||
"../modules/rtp_rtcp",
|
||||
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
||||
"../modules/video_coding:webrtc_h264",
|
||||
"../modules/video_coding:webrtc_vp8",
|
||||
"../modules/video_coding:webrtc_vp9",
|
||||
"../system_wrappers",
|
||||
"../video",
|
||||
"../voice_engine",
|
||||
"//testing/gmock",
|
||||
"//testing/gtest",
|
||||
]
|
||||
@ -561,9 +515,6 @@ rtc_source_set("test_renderer") {
|
||||
|
||||
deps = [
|
||||
":test_support",
|
||||
"..:webrtc_common",
|
||||
"../base:rtc_base_approved",
|
||||
"../common_video",
|
||||
"../modules/media_file",
|
||||
"//testing/gtest",
|
||||
]
|
||||
@ -580,10 +531,7 @@ rtc_source_set("audio_codec_mocks") {
|
||||
]
|
||||
|
||||
deps = [
|
||||
":test_support",
|
||||
"../api/audio_codecs:audio_codecs_api",
|
||||
"../api/audio_codecs:builtin_audio_decoder_factory",
|
||||
"../base:rtc_base_approved",
|
||||
"//testing/gmock",
|
||||
]
|
||||
}
|
||||
|
||||
@ -15,7 +15,6 @@ rtc_static_library("webrtc_fuzzer_main") {
|
||||
"webrtc_fuzzer_main.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../system_wrappers:field_trial_default",
|
||||
"../../system_wrappers:metrics_default",
|
||||
"//testing/libfuzzer:libfuzzer_main",
|
||||
@ -65,7 +64,6 @@ webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
|
||||
"vp8_qp_parser_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../modules/video_coding:video_coding_utility",
|
||||
"../../modules/video_coding/",
|
||||
]
|
||||
}
|
||||
@ -75,7 +73,6 @@ webrtc_fuzzer_test("h264_bitstream_parser_fuzzer") {
|
||||
"h264_bitstream_parser_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../common_video",
|
||||
"../../modules/video_coding/",
|
||||
]
|
||||
}
|
||||
@ -85,7 +82,6 @@ webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
|
||||
"flexfec_header_reader_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
]
|
||||
}
|
||||
@ -96,7 +92,6 @@ webrtc_fuzzer_test("flexfec_sender_fuzzer") {
|
||||
]
|
||||
deps = [
|
||||
"../../modules/rtp_rtcp",
|
||||
"../../system_wrappers",
|
||||
]
|
||||
libfuzzer_options = [ "max_len=200" ]
|
||||
}
|
||||
@ -106,7 +101,6 @@ webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
|
||||
"ulpfec_header_reader_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
"../../modules/rtp_rtcp:fec_test_helper",
|
||||
]
|
||||
@ -128,7 +122,6 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
|
||||
"flexfec_receiver_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
]
|
||||
libfuzzer_options = [ "max_len=2000" ]
|
||||
@ -140,7 +133,6 @@ webrtc_fuzzer_test("packet_buffer_fuzzer") {
|
||||
]
|
||||
deps = [
|
||||
"../../modules/video_coding/",
|
||||
"../../system_wrappers",
|
||||
]
|
||||
libfuzzer_options = [ "max_len=2000" ]
|
||||
}
|
||||
@ -150,9 +142,7 @@ webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
|
||||
"rtcp_receiver_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
"../../system_wrappers:system_wrappers",
|
||||
]
|
||||
seed_corpus = "corpora/rtcp-corpus"
|
||||
}
|
||||
@ -181,11 +171,8 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
|
||||
"congestion_controller_feedback_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../logging:rtc_event_log_api",
|
||||
"../../logging:rtc_event_log_impl",
|
||||
"../../modules/congestion_controller/",
|
||||
"../../modules/remote_bitrate_estimator:remote_bitrate_estimator",
|
||||
"../../modules/rtp_rtcp",
|
||||
]
|
||||
}
|
||||
|
||||
@ -194,12 +181,6 @@ rtc_static_library("audio_decoder_fuzzer") {
|
||||
"audio_decoder_fuzzer.cc",
|
||||
"audio_decoder_fuzzer.h",
|
||||
]
|
||||
deps = [
|
||||
"../..:webrtc_common",
|
||||
"../../api/audio_codecs:audio_codecs_api",
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
]
|
||||
}
|
||||
|
||||
webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
|
||||
@ -279,7 +260,6 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
|
||||
"../../base:rtc_base_approved",
|
||||
"../../base:rtc_base_tests_utils",
|
||||
"../../modules/audio_coding:neteq",
|
||||
"../../modules/audio_coding:neteq_test_tools",
|
||||
"../../modules/audio_coding:neteq_tools_minimal",
|
||||
"../../modules/audio_coding:pcm16b",
|
||||
"../../modules/rtp_rtcp",
|
||||
@ -333,7 +313,6 @@ webrtc_fuzzer_test("pseudotcp_parser_fuzzer") {
|
||||
"pseudotcp_parser_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base",
|
||||
"../../p2p:rtc_p2p",
|
||||
]
|
||||
}
|
||||
@ -343,7 +322,6 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") {
|
||||
"transport_feedback_packet_loss_tracker_fuzzer.cc",
|
||||
]
|
||||
deps = [
|
||||
"../../base:rtc_base_approved",
|
||||
"../../modules/rtp_rtcp",
|
||||
"../../voice_engine",
|
||||
]
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user