| commit | 7203caf1dc5de35ff9b59f942bdea5f85efbe7c7 | [log] [tgz] |
|---|---|---|
| author | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | Thu Oct 11 11:54:31 2012 |
| committer | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | Thu Oct 11 11:54:31 2012 |
| tree | 8fda5c76af6eea034d5f316bd1b4368eeb284e53 | |
| parent | 63f3e1e79cdb3816b609ab27a112e3e0beb48243 [diff] |
Ensures that we always run the low-latency audio capture at natively 128 audio frames. A FIFO is used to adapt to the buffer size requested by the client. Tested with WebRTC clients in Chrome as well. Added media_unittests as well for different sample rates. BUG=154352 TEST=content_unittests --v=1 --gtest_filter=WebRTC* Review URL: https://chromiumcodereview.appspot.com/11099013 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@161328 0039d316-1c4b-4281-b951-d872f2087c98