Set WEBRTC_USE_H264 in common_config
Bug: none Change-Id: I3dce8fdc409c88cdd771ea57eca3ea375e6e82c9 Reviewed-on: https://webrtc-review.googlesource.com/c/121948 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26603}
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commit
b76b9ba8ce
4
BUILD.gn
4
BUILD.gn
@ -242,6 +242,10 @@ config("common_config") {
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defines += [ "ENABLE_EXTERNAL_AUTH" ]
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defines += [ "ENABLE_EXTERNAL_AUTH" ]
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}
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}
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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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if (build_with_chromium) {
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if (build_with_chromium) {
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defines += [
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defines += [
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# NOTICE: Since common_inherited_config is used in public_configs for our
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# NOTICE: Since common_inherited_config is used in public_configs for our
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@ -481,10 +481,6 @@ if (rtc_include_tests) {
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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deps += [
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deps += [
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":rtc_internal_video_codecs",
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":rtc_internal_video_codecs",
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":rtc_media",
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":rtc_media",
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@ -587,10 +583,6 @@ if (rtc_include_tests) {
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sources += [ "sctp/sctp_transport_unittest.cc" ]
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sources += [ "sctp/sctp_transport_unittest.cc" ]
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}
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}
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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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if (rtc_opus_support_120ms_ptime) {
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if (rtc_opus_support_120ms_ptime) {
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defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
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defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=1" ]
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} else {
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} else {
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@ -336,7 +336,6 @@ rtc_static_library("webrtc_h264") {
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if (rtc_use_h264) {
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if (rtc_use_h264) {
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assert(!(is_win && !is_clang), "See: bugs.webrtc.org/9213#c13.")
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assert(!(is_win && !is_clang), "See: bugs.webrtc.org/9213#c13.")
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defines += [ "WEBRTC_USE_H264" ]
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sources += [
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sources += [
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"codecs/h264/h264_color_space.cc",
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"codecs/h264/h264_color_space.cc",
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"codecs/h264/h264_color_space.h",
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"codecs/h264/h264_color_space.h",
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@ -761,9 +760,6 @@ if (rtc_include_tests) {
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rtc_source_set("video_coding_modules_tests") {
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rtc_source_set("video_coding_modules_tests") {
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testonly = true
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testonly = true
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defines = []
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defines = []
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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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sources = [
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sources = [
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"codecs/h264/test/h264_impl_unittest.cc",
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"codecs/h264/test/h264_impl_unittest.cc",
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@ -322,9 +322,6 @@ if (rtc_include_tests) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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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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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_use_h264) {
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defines = [ "WEBRTC_USE_H264" ]
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}
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}
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}
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rtc_executable("video_loopback") {
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rtc_executable("video_loopback") {
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@ -584,9 +581,6 @@ if (rtc_include_tests) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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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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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_use_h264) {
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defines += [ "WEBRTC_USE_H264" ]
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}
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if (!build_with_mozilla) {
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if (!build_with_mozilla) {
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deps += [ "../media:rtc_media_base" ]
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deps += [ "../media:rtc_media_base" ]
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}
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}
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