| commit | 7203caf1dc5de35ff9b59f942bdea5f85efbe7c7 | |
| author | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | |
| Thu, 11 Oct 2012 11:54:31 +0000 (11 11:54 +0000) | ||
| committer | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | |
| Thu, 11 Oct 2012 11:54:31 +0000 (11 11:54 +0000) | ||
| tree | 8fda5c76af6eea034d5f316bd1b4368eeb284e53 | treesnapshot (tar.gz zip) |
| parent | 63f3e1e79cdb3816b609ab27a112e3e0beb48243 | commitdiff |
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
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