| commit | f4dd191b288830be57787b547c05094652c2171c | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Thu Jun 15 01:55:38 2017 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jun 15 08:55:38 2017 +0000 |
| tree | 5f533a6235a7673e9b2665a50b3364e68ef3725e | |
| parent | af66f2ca8a3609184518af0d5083f9ee492ec20b [diff] |
Change existing aec dump tests to use webrtc::AecDump.
Currently the debug dump functionality of WebRTC (a log of all
AudioProcessing operations) was tested by the following tests:
1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing
from a debug dump, and verifies that the same debug dump is
recorded.
2. DebugDumpTest.* which is a comprehensive test of the debug dump
operations. AudioProcessing configuration is changed, and the dump
is scanned for the change.
3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that
debug dumping can be started and files written.
This CL replaces the debug dump mechanism in all these tests to
webrtc::AecDump. Some of the tests are adapted to the chenges of the
new API to AecDump {Start,Stop}DebugRecording: the old functions
signal errors when a file cannot be opened. With AecDump, the
AecDumpFactory instead returns a nullptr.
The CL also changes audioproc_f to use AecDump.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2864373002
Cr-Commit-Position: refs/heads/master@{#18605}
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.