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https://github.com/mollyim/webrtc.git
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aa42ecde9a
This allows clients to exclude the transient suppression submodule from WebRTC builds, by defining WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR. The changes have been shown to be bitexact for a test dataset (when the flag is _not_ defined.) No-Try: True Bug: webrtc:11226, webrtc:11292 Change-Id: I6931c82a280a9b40a53ee1c2a9820ed9e674a9a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171421 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30978}
116 lines
5 KiB
Markdown
116 lines
5 KiB
Markdown
# API header files
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As a user of the WebRTC library, you may use headers and build files
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in the following directories:
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API directory | Including subdirectories?
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--------------|-------------------------
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`api` | Yes
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For now, you may also use headers and build files in the following
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legacy API directories—but see the
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[disclaimer](#legacy-disclaimer) below.
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Legacy API directory | Including subdirectories?
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-------------------------------------------|--------------------------
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`common_audio/include` | No
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`media/base` | No
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`media/engine` | No
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`modules/audio_coding/include` | No
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`modules/audio_device/include` | No
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`modules/audio_processing/include` | No
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`modules/bitrate_controller/include` | No
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`modules/congestion_controller/include` | No
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`modules/include` | No
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`modules/remote_bitrate_estimator/include` | No
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`modules/rtp_rtcp/include` | No
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`modules/rtp_rtcp/source` | No
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`modules/utility/include` | No
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`modules/video_coding/codecs/h264/include` | No
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`modules/video_coding/codecs/vp8/include` | No
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`modules/video_coding/codecs/vp9/include` | No
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`modules/video_coding/include` | No
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`pc` | No
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`rtc_base` | No
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`system_wrappers/include` | No
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While the files, types, functions, macros, build targets, etc. in the
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API and legacy API directories will sometimes undergo incompatible
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changes, such changes will be announced in advance to
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[discuss-webrtc@googlegroups.com][discuss-webrtc], and a migration
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path will be provided.
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[discuss-webrtc]: https://groups.google.com/forum/#!forum/discuss-webrtc
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In the directories not listed in the tables above, incompatible
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changes may happen at any time, and are not announced.
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## <a name="legacy-disclaimer"></a>The legacy API directories contain some things you shouldn’t use
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The legacy API directories, in addition to things that genuinely
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should be part of the API, also contain things that should *not* be
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part of the API. We are in the process of moving the good stuff to the
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`api` directory tree, and will remove directories from the legacy list
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once they no longer contain anything that should be in the API.
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In other words, if you find things in the legacy API directories that
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don’t seem like they belong in the WebRTC native API,
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don’t grow too attached to them.
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## All these worlds are yours—except Europa
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In the API headers, or in files included by the API headers, there are
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types, functions, namespaces, etc. that have `impl` or `internal` in
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their names (in various styles, such as `CamelCaseImpl`,
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`snake_case_impl`). They are not part of the API, and may change
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incompatibly at any time; do not use them.
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# Preprocessor macros
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The following preprocessor macros are read (but never set) by WebRTC; they allow
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you to enable or disable parts of WebRTC at compile time.
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Be sure to set them the same way in all translation units that include WebRTC
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code.
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## `WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS`
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If you want to ship your own set of SSL certificates and inject them into WebRTC
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PeerConnections, you will probably want to avoid to compile and ship WebRTC's
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default set of SSL certificates.
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You can achieve this by defining the preprocessor macro
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`WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS`. If you use GN, you can just set the GN
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argument `rtc_builtin_ssl_root_certificates` to false and GN will define the
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macro for you.
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## `WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT`
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If you want to provide your own implementation of `webrtc::field_trial` functions
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(more info [here][field_trial_h]) you will have to exclude WebRTC's default
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implementation.
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You can achieve this by defining the preprocessor macro
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`WEBRTC_EXCLUDE_FIELD_TRIAL_DEFAULT`. If you use GN, you can just set the GN
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argument `rtc_exclude_field_trial_default` to true and GN will define the
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macro for you.
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[field_trial_h]: https://webrtc.googlesource.com/src/+/master/system_wrappers/include/field_trial.h
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## `WEBRTC_EXCLUDE_METRICS_DEFAULT`
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If you want to provide your own implementation of `webrtc::metrics` functions
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(more info [here][metrics_h]) you will have to exclude WebRTC's default
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implementation.
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You can achieve this by defining the preprocessor macro
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`WEBRTC_EXCLUDE_METRICS_DEFAULT`. If you use GN, you can just set the GN
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argument `rtc_exclude_metrics_default` to true and GN will define the
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macro for you.
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[metrics_h]: https://webrtc.googlesource.com/src/+/master/system_wrappers/include/metrics.h
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## `WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR`
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The transient suppressor functionality in the audio processing module is not
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always used. If you wish to exclude it from the build in order to preserve
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binary size, then define the preprocessor macro
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`WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR`. If you use GN, you can just set the GN
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argument `rtc_exclude_transient_suppressor` to true and GN will define the macro
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for you.
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