bfde543f73
This will allow us to fix the sample rate of each AudioDecoder at instantiation time. This change results in different checksums for the following tests: AcmReceiverBitExactnessOldApi.8kHzOutput AcmReceiverBitExactnessOldApi.16kHzOutput AcmReceiverBitExactnessOldApi.32kHzOutput AcmReceiverBitExactnessOldApi.48kHzOutputExternalDecoder AcmReceiverBitExactnessOldApi.48kHzOutput Because they make an ACM and then ask it to decode both 16 kHz and 32 kHz iSAC. (The arm32 and arm64 checksums didn't change, because the tests skip 32 kHz iSAC on arm.) BUG=webrtc:5801 Review URL: https://codereview.webrtc.org/1908923002 Cr-Commit-Position: refs/heads/master@{#12463}
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. This page is maintained by the Google Chrome team.
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