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author | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-02-06 08:41:24 +0000 |
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committer | henrika@chromium.org <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-02-06 08:41:24 +0000 |
commit | 3ca01de32c4b1e40d45d2daeedcb841429ce4704 (patch) | |
tree | 826de070a239aeca2eb3c37392eabd3560f674e3 /content/browser/device_orientation | |
parent | 2987c2f633fdff2ba80ac2894384698c850948dd (diff) | |
download | chromium_src-3ca01de32c4b1e40d45d2daeedcb841429ce4704.zip chromium_src-3ca01de32c4b1e40d45d2daeedcb841429ce4704.tar.gz chromium_src-3ca01de32c4b1e40d45d2daeedcb841429ce4704.tar.bz2 |
Avoids irregular OnMoreData callbacks on Windows using Core Audio.
Browser changes:
- Improves how native audio buffer sizes are derived on Windows.
- Forces user to always open up at native audio paramters.
- Improved internal scheme to set up the actial endpoint buffer based on input size.
- Refactored WSAPI output implementation and introduced CoreAudioUtil methods.
- Harmonized WSAPI output implementation with exusting unified implementation (to prepare for future merge).
- Changed GetAudioHardwareBufferSize() in audio_util.
Render changes for WebRTC:
- WebRTC now always asks for an output stream using native parameters to avoid rebuffering in the audio converter.
- Any buffer-size mismatch is now taken care of in WebRtcAudioRendrer using a pull FIFO. Delay estimates are also compensated if FIFO is used.
- Added DCHECKs to verify that methods are called on the expected threads.
BUG=170498
TEST=media_unittests, content_unittests, HTML5 audio tests in Chrome, WebAudio and Flash tests in Chrome, WebRTC tests in Chrome.
Review URL: https://codereview.chromium.org/12049070
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@180936 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/browser/device_orientation')
0 files changed, 0 insertions, 0 deletions