9868042b05
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc