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| author | dalecurtis@chromium.org <dalecurtis@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-10-01 03:50:03 +0000 |
|---|---|---|
| committer | dalecurtis@chromium.org <dalecurtis@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2013-10-01 03:50:03 +0000 |
| commit | 2fb4b740a8f6c79ae376c5f0257ed66ac3ef4246 (patch) | |
| tree | de534a2cb59ad9c9323923095150f8ea8a08f1bc /content/renderer | |
| parent | e8475f8ac4fa50edffd37396c0cb386e3e817879 (diff) | |
| download | chromium_src-2fb4b740a8f6c79ae376c5f0257ed66ac3ef4246.zip chromium_src-2fb4b740a8f6c79ae376c5f0257ed66ac3ef4246.tar.gz chromium_src-2fb4b740a8f6c79ae376c5f0257ed66ac3ef4246.tar.bz2 | |
Remove the last vestiges of audio_util.
- Removes GetHighLatencyOutputBufferSize() since we no longer
have any high latency users. Fallback uses a minimum 2048
buffer size in AudioOutputResampler.
- Moves NumberOfWaveOutBuffers() to AudioManagerWin. Documents
existing broken unit test edge case for Vista.
- Moves GetUserBufferSize() to AudioManagerBase as a protected
method.
BUG=120319
TEST=compiles
Review URL: https://codereview.chromium.org/25133002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@226131 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'content/renderer')
| -rw-r--r-- | content/renderer/media/webrtc_audio_capturer.cc | 1 | ||||
| -rw-r--r-- | content/renderer/media/webrtc_audio_device_impl.cc | 1 |
2 files changed, 0 insertions, 2 deletions
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc index bd1af6a..4b9b503 100644 --- a/content/renderer/media/webrtc_audio_capturer.cc +++ b/content/renderer/media/webrtc_audio_capturer.cc @@ -12,7 +12,6 @@ #include "content/renderer/media/audio_device_factory.h" #include "content/renderer/media/webrtc_audio_device_impl.h" #include "content/renderer/media/webrtc_local_audio_track.h" -#include "media/audio/audio_util.h" #include "media/audio/sample_rates.h" namespace content { diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc index 0692ddc..aadcda9 100644 --- a/content/renderer/media/webrtc_audio_device_impl.cc +++ b/content/renderer/media/webrtc_audio_device_impl.cc @@ -12,7 +12,6 @@ #include "content/renderer/media/webrtc_audio_renderer.h" #include "content/renderer/render_thread_impl.h" #include "media/audio/audio_parameters.h" -#include "media/audio/audio_util.h" #include "media/audio/sample_rates.h" using media::AudioParameters; |