8789376cd3
Move ChannelBuffer class to channel_buffer file
...
No change in functionallity.
BUG=webrtc:3146
R=andrew@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-27 23:40:25 +00:00
641bda6f9c
Initialize ChannelBuffer's memory to avoid uninitialized reads.
...
Removed the zero out memset in this change:
https://review.webrtc.org/24469004/
assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.
TBR=claguna@google.com
Review URL: https://webrtc-codereview.appspot.com/28429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 23:11:44 +00:00
17454f79dc
Add ctors to ChannelBuffer to enable copying on construction.
...
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.
R=claguna@google.com
Review URL: https://webrtc-codereview.appspot.com/24469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:27:04 +00:00
e364ac902f
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
...
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)
R=aluebs@webrtc.org , andrew@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-18 07:50:29 +00:00
af93fc08a1
AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter
...
R=aluebs@webrtc.org , minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13019004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 08:18:33 +00:00
f2aafe4355
Added include of assert.h for files calling assert but missing the include.
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:54:17 +00:00
ddbb8a2c24
Support arbitrary input/output rates and downmixing in AudioProcessing.
...
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:00:04 +00:00
