| commit | 9868042b053d52c45c632f09c1d22cb13bc1ad41 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Thu Aug 31 13:47:32 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Aug 31 13:47:32 2017 |
| tree | dfd2f438a2de48aa12499ca9b857b74bd58263c4 | |
| parent | 137cd1d788ea2630d5996d8e76cdcae410f014fb [diff] |
Removes unused APIs from the ADM (part II). Removes: int32_t SpeakerVolumeStepSize(uint16_t* stepSize) int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) int32_t MicrophoneBoostIsAvailable(bool* available) int32_t SetMicrophoneBoost(bool enable) int32_t MicrophoneBoost(bool* enabled) int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0) int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) int32_t CPULoad(uint16_t* load) int32_t StartRawOutputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize]) int32_t StopRawOutputFileRecording() int32_t StartRawInputFileRecording(const char pcmFileNameUTF8[kAdmMaxFileNameSize]) int32_t StopRawInputFileRecording() int32_t ResetAudioDevice() BUG=webrtc:7306 Review-Url: https://codereview.webrtc.org/3006803002 Cr-Commit-Position: refs/heads/master@{#19632}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.