Commit Graph

15 Commits

Author SHA1 Message Date
4161715e3f Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

BUG=
R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00
8a2c84f59d Log the Android Audio API choice correctly.
BUG=3699
TEST=Manual Test
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6915 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-18 03:02:42 +00:00
122caa51b1 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue.
CL also replaces deprecated AudioSession calls with equivalent AVAudioSession ones.

BUG=3487
R=glaznev@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 20:20:47 +00:00
8454ad1b3e Reland: Making WebRTC able to play and record audio to files for tests.
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6403 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 14:12:04 +00:00
e08a11c4a1 Revert 6395 "Making WebRTC able to play and record audio to file..."
> Making WebRTC able to play and record audio to files for tests.
> 
> By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
> WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
> play out audio to a file and feed audio in from a file. We want to do
> so we can better test WebRTC-using applications by recording what the
> audio stack outputs and feeding known audio in for quality tests.
> 
> R=henrika@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/20609004

TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6396 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 10:40:30 +00:00
fa042ca15d Making WebRTC able to play and record audio to files for tests.
By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6395 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11 09:57:23 +00:00
c7c432aa9b Remove AudioDevice::{Microphone,Speaker}IsAvailable.
This was only used for logging, except on Mac, where the methods are
now private.

BUG=3132
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-02 16:49:26 +00:00
0e65fdaa3b Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.
BUG=chromium:346399
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10139004

Patch from Peter Kasting <pkasting@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-21 10:26:42 +00:00
9ee75e9c77 Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
7ae8495779 Removed unnecessary Pulse init from VoE startup.
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 21:01:34 +00:00
811269df40 Include files from webrtc/.. paths in audio_device/.
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 13:24:38 +00:00
2550988baa WebRtc_Word32 -> int32_t in audio_device/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1302006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3793 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 10:30:35 +00:00
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00