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RTC configuration values set via webrtc.gni

The build system uses those configuration values to determine which #define to declare.

Configuration Values

rtc_enable_symbol_export = true

Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h) expand to code that will manage symbols visibility.

#define WEBRTC_ENABLE_SYMBOL_EXPORT

rtc_enable_objc_symbol_export = true

Setting this to true will make RTC_OBJC_EXPORT expand to code that will manage symbols visibility. By default, Obj-C/Obj-C++ symbols are exported if C++ symbols are but setting this arg to true while keeping rtc_enable_symbol_export= false will only export RTC_OBJC_EXPORT annotated symbols.

#define WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT

rtc_dlog_always_on = false

Setting this to true, will make RTC_DLOG() expand to log statements instead of being removed by the preprocessor. This is useful for example to be able to get RTC_DLOGs on a release build.

 #define DLOG_ALWAYS_ON

rtc_enable_google_benchmarks = false

Enables additional build targets that rely on //third_party/google_benchmarks.

rtc_exclude_field_trial_default = false

Setting this to true will define WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT which will tell the pre-processor to remove the default definition of symbols needed to use field_trial. In that case a new implementation needs to be provided.

When WebRTC is built as part of Chromium it should exclude the default implementation of field_trial unless it is building for NACL or Chromecast.

rtc_exclude_metrics_default = false

Setting this to true will define WEBRTC_EXCLUDE_METRICS_DEFAULT which will tell the pre-processor to remove the default definition of symbols needed to use metrics. In that case a new implementation needs to be provided.

rtc_exclude_system_time = false

Setting this to true will define WEBRTC_EXCLUDE_SYSTEM_TIME which will tell the pre-processor to remove the default definition of the SystemTimeNanos() which is defined in rtc_base/system_time.cc. In that case a new implementation needs to be provided.

rtc_builtin_ssl_root_certificates = true

Setting this to false will require the API user to pass in their own SSLCertificateVerifier to verify the certificates presented from a TLS-TURN server. In return disabling this saves around 100kb in the binary.

#define WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS

rtc_include_ilbc = true

Include the iLBC audio codec?

rtc_include_opus = true

Disable this to avoid building the Opus audio codec.

rtc_opus_support_120ms_ptime = true

Enable this if the Opus version upon which WebRTC is built supports direct encoding of 120 ms packets.

rtc_opus_variable_complexity = false

Enable this to let the Opus audio codec change complexity on the fly.

rtc_enable_external_auth = false

Enable when an external authentication mechanism is used for performing packet authentication for RTP packets instead of libsrtp.

#ifdef ENABLE_EXTERNAL_AUTH
#undef ENABLE_EXTERNAL_AUTH
#endif

apm_debug_dump = false

Selects whether debug dumps for the audio processing module should be generated.

rtc_exclude_audio_processing_module = false

Selects whether the audio processing module should be excluded.

#ifdef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
#undef WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE
#endif

rtc_enable_bwe_test_logging = false

Set this to true to enable BWE test logging.

rtc_build_examples = false

Set this to true to build examples.

rtc_build_tools = true

Set this to false to skip building tools.

rtc_use_x11 = false

Set this to false to skip building code that requires X11.

rtc_use_pipewire = ‘false’

Set this to use PipeWire on the Wayland display server. By default it’s only enabled on desktop Linux (excludes ChromeOS) and only when using the sysroot as PipeWire is not available in older and supported Ubuntu and Debian distributions.

rtc_link_pipewire = false

Set this to link PipeWire and required libraries directly instead of using the dlopen.

build_with_mozilla = false

Enable to use the Mozilla internal settings.

rtc_enable_android_aaudio = false

Experimental: enable use of Android AAudio which requires Android SDK 26 or above and NDK r16 or above.

rtc_sanitize_coverage = ""

Set to func, block, edge for coverage generation. At unit test runtime set UBSAN_OPTIONS="coverage=1". It is recommend to set include_examples=0. Use llvm’s sancov -html-report for human readable reports. See http://clang.llvm.org/docs/SanitizerCoverage.html .

rtc_prefer_fixed_point = current_cpu == arm || current_cpu == arm64

Selects fixed-point code where possible (true for arm and arm64).

rtc_build_with_neon = true

Determines whether NEON code will be built.

rtc_use_h264 = false

Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on all platforms except Android and iOS.

#ifdef WEBRTC_USE_H264
#undef WEBRTC_USE_H264
#endif

rtc_use_absl_mutex = false

Enable this flag to make webrtc::Mutex be implemented by absl::Mutex.

#ifdef WEBRTC_ABSL_MUTEX
#undef WEBRTC_ABSL_MUTEX
#endif

rtc_ios_use_opengl_rendering = is_ios && target_environment != catalyst

Determines whether OpenGL is available on iOS.

rtc_include_builtin_audio_codecs = true

When set to false, builtin audio encoder/decoder factories and all the audio codecs they depend on will not be included in libwebrtc.{a|lib} (they will still be included in libjingle_peerconnection_so.so and WebRTC.framework)

rtc_include_dav1d_in_internal_decoder_factory = false

Includes the dav1d decoder in the internal decoder factory when set to true.

#define RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY

rtc_strict_field_trials = ""

When enabled, a run-time check will make sure that all field trial keys have been registered in accordance with the field trial policy, see g3doc/field-trials.md. The value can be set to the following:

  • “dcheck”

RTC_DCHECKs that the field trial has been registered. RTC_DCHECK must be enabled separately.

  • “warn”

RTC_LOGs a message with LS_WARNING severity if the field trial hasn’t been registered.

#define WEBRTC_STRICT_FIELD_TRIALS 0 // ""
#define WEBRTC_STRICT_FIELD_TRIALS 1 // "dcheck"
#define WEBRTC_STRICT_FIELD_TRIALS 2 // "warn"

rtc_enable_protobuf = true

Enables the use of protocol buffers for debug recordings.

#define WEBRTC_ENABLE_PROTOBUF 1

rtc_enable_sctp = true

Set this to disable building with support for SCTP data channels.

#define WEBRTC_HAVE_SCTP

rtc_build_json = true

rtc_build_libsrtp = true

rtc_build_libvpx = true

rtc_libvpx_build_vp9 = true

#define RTC_ENABLE_VP9

rtc_build_opus = true

rtc_build_ssl = true

Disable these to not build components which can be externally provided.

rtc_enable_libevent = false

rtc_build_libevent = false

Enable libevent task queues on platforms that support it.

rtc_include_pulse_audio = false

Excluded in Chromium since its prerequisites don’t require Pulse Audio.

rtc_include_internal_audio_device = true

Chromium uses its own IO handling, so the internal ADM is only built for standalone WebRTC.

#define WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE

rtc_enable_avx2 = false

Set this to true to enable the avx2 support in webrtc.

#define WEBRTC_ENABLE_AVX2

rtc_include_tests = false

Set this to true to build the unit tests. Disabled when building with Chromium or Mozilla.

#define WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS 1

rtc_use_x11_extensions = false

Set this to false to skip building code that also requires X11 extensions such as Xdamage, Xfixes.

rtc_disable_logging = false

Set this to true to fully remove logging from WebRTC.

#ifdef RTC_DISABLE_LOGGING
#undef RTC_DISABLE_LOGGING
#endif

rtc_disable_trace_events = false

Set this to true to disable trace events.

#ifdef RTC_DISABLE_TRACE_EVENTS
#undef RTC_DISABLE_TRACE_EVENTS
#endif

rtc_disable_check_msg = false

Set this to true to disable detailed error message and logging for RTC_CHECKs.

#define RTC_DISABLE_CHECK_MSG

rtc_disable_metrics = false

Set this to true to disable webrtc metrics.

#ifdef RTC_DISABLE_METRICS
#undef RTC_DISABLE_METRICS
#endif

rtc_exclude_transient_suppressor = false

Set this to true to exclude the transient suppressor in the audio processing module from the build.

#ifdef WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR
#undef WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR
#endif

rtc_desktop_capture_supported = is_mac

Desktop capturer is supported only on Windows, OSX and Linux.

Additional defines based on BUILDCONFIG.gn

is_ios = current_os == ios

#define WEBRTC_IOS
#define WEBRTC_MAC

is_mac = current_os == mac

#define WEBRTC_MAC

is_posix = !is_win && !is_fuchsia

#define WEBRTC_POSIX

current_cpu == arm64

#define WEBRTC_ARCH_ARM64
#define WEBRTC_HAS_NEON

current_cpu == arm

#define WEBRTC_ARCH_ARM
#define WEBRTC_HAS_NEON

arm_version >= 7

#define WEBRTC_ARCH_ARM_V7

arm_use_neon == true

#define WEBRTC_HAS_NEON