| commit | acc25ac1e26010c501fd98d59738e95e22a7015c | [log] [tgz] |
|---|---|---|
| author | mcasas <mcasas@chromium.org> | Sat Jan 30 00:29:45 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Sat Jan 30 00:30:31 2016 |
| tree | 4137e5276a78dbeaf71c0d0614ca660717be2a36 | |
| parent | b3abb79f6131f01f363dc7bea511ced207019c4f [diff] |
MediaRecorder: support sampling rate adaption in AudioTrackRecorder
This CL adds support for audio coming from the MediaStream
Tracks with a sampling rate not directly supported by Opus
({48k, 24k, 16k, 12k, 8k}samples/s) by adding a AudioConverter
in AudioTrackRecorder::AudioEncoder.
BUG=569089
TEST =
- unit tests, which are beefed up
- content_browsertests
- demo [1]
- corrected bug's file [2]
(correction being: s/opus/webm/ for the mimeType to
use for capture.)
I used a variety of input wav's, mostly the original
sfx5.wav in the bug and the "miscellaneous" entries in [3]
[1] https://webrtc.github.io/samples/src/content/getusermedia/record/
[2] https://rawgit.com/Miguelao/demos/master/audiotranscode.html
[3] http://download.wavetlan.com/SVV/Media/HTTP/http-wav.htm
Review URL: https://codereview.chromium.org/1579693006
Cr-Commit-Position: refs/heads/master@{#372491}