As a user of the WebRTC library, you may use headers and build files in the following directories:
| API directory | Including subdirectories? |
|---|---|
api |
Yes |
For now, you may also use headers and build files in the following legacy API directories—but see the disclaimer below.
| Legacy API directory | Including subdirectories? |
|---|---|
common_audio/include |
No |
media/base |
No |
media/engine |
No |
modules/audio_coding/include |
No |
modules/audio_device/include |
No |
modules/audio_processing/include |
No |
modules/bitrate_controller/include |
No |
modules/congestion_controller/include |
No |
modules/include |
No |
modules/remote_bitrate_estimator/include |
No |
modules/rtp_rtcp/include |
No |
modules/rtp_rtcp/source |
No |
modules/utility/include |
No |
modules/video_coding/codecs/h264/include |
No |
modules/video_coding/codecs/vp8/include |
No |
modules/video_coding/codecs/vp9/include |
No |
modules/video_coding/include |
No |
pc |
No |
rtc_base |
No |
system_wrappers/include |
No |
While the files, types, functions, macros, build targets, etc. in the API and legacy API directories will sometimes undergo incompatible changes, such changes will be announced in advance to discuss-webrtc@googlegroups.com, and a migration path will be provided.
In the directories not listed in the tables above, incompatible changes may happen at any time, and are not announced.
The legacy API directories, in addition to things that genuinely
should be part of the API, also contain things that should not be
part of the API. We are in the process of moving the good stuff to the
api directory tree, and will remove directories from the legacy list
once they no longer contain anything that should be in the API.
In other words, if you find things in the legacy API directories that don’t seem like they belong in the WebRTC native API, don’t grow too attached to them.
In the API headers, or in files included by the API headers, there are
types, functions, namespaces, etc. that have impl or internal in
their names (in various styles, such as CamelCaseImpl,
snake_case_impl). They are not part of the API, and may change
incompatibly at any time; do not use them.
The following preprocessor macros are read (but never set) by WebRTC; they allow you to enable or disable parts of WebRTC at compile time.
Be sure to set them the same way in all translation units that include WebRTC code.
If you want to ship your own set of SSL certificates and inject them into WebRTC PeerConnections, you will probably want to avoid to compile and ship WebRTC's default set of SSL certificates.
You can achieve this by defining the preprocessor macro
WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS. If you use GN, you can just set the GN
argument rtc_builtin_ssl_root_certificates to false and GN will define the
macro for you.
If you want to provide your own implementation of webrtc::field_trial functions
(more info here) you will have to exclude WebRTC's default
implementation.
You can achieve this by defining the preprocessor macro
WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT. If you use GN, you can just set the GN
argument rtc_exclude_field_trial_default to true and GN will define the
macro for you.
If you want to provide your own implementation of webrtc::metrics functions
(more info here) you will have to exclude WebRTC's default
implementation.
You can achieve this by defining the preprocessor macro
WEBRTC_EXCLUDE_METRICS_DEFAULT. If you use GN, you can just set the GN
argument rtc_exclude_metrics_default to true and GN will define the
macro for you.
The transient suppressor functionality in the audio processing module is not
always used. If you wish to exclude it from the build in order to preserve
binary size, then define the preprocessor macro
WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR. If you use GN, you can just set the GN
argument rtc_exclude_transient_suppressor to true and GN will define the macro
for you.