| commit | 2a4906532f9de4b247349af6f8f75d8931fb9e38 | [log] [tgz] |
|---|---|---|
| author | henrika <henrika@webrtc.org> | Tue Aug 28 13:52:10 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 29 08:43:03 2018 |
| tree | d872e2359fc34da27e122b1348c22f6c8b5afaf2 | |
| parent | f18b35284288ac851b77db88df3e2e8d2273db97 [diff] |
Increases max size of webrtc::AudioFrame from 60ms to 120ms @32kHz.
Existing max size seems a bit random imho. THis CL extends it from 60ms
to 120ms but the actual goal is to allow usage of 20ms @192kHz since
that is the largest possible sample rate which can be selected on most
platforms.
Recent work on the ADM for Windows ensures that the ADM now supports
192kHz.
Without this change, we will hit DCHECK:s like these:
RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
AudioFrame::kMaxDataSizeBytes)
when 192kHz is utilized.
Bug: webrtc:9265
Change-Id: Ib4f76a2ecfb1a541776938b8eed801ad64386daa
Reviewed-on: https://webrtc-review.googlesource.com/96542
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24473}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.
Authoritative list of directories that contain the native API header files.