Commit Graph

11 Commits

Author SHA1 Message Date
e7edea9759 Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #5 id:80001 of https://codereview.chromium.org/2037623002/ )
Reason for revert:
voice_engine_unittests: FilePlayerTest.PlayWavPcm16File and FilePlayerTest.PlayWavPcmuFile fail on 32-bit android (android_rel and android-dbg try bots, Android32 Tests (L Nexus5) and Android32 Tests (L Nexus7.2) build bots).

Not sure why this would happen, since I just moved the test without modifying it. Some test filtering that no longer manages to disable them? Anyway, reverting until I know how to fix.

This was actually caught by the try bots, but I missed it because I was manually ignoring them because of an error with the bots. :-(

Original issue's description:
> Move FilePlayer and FileRecorder to Voice Engine
>
> Because Voice Engine was the only user.
>
> R=perkj@webrtc.org, solenberg@webrtc.org
>
> Committed: 65874b163e

TBR=perkj@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2092633002
Cr-Commit-Position: refs/heads/master@{#13267}
2016-06-22 23:29:58 +00:00
65874b163e Move FilePlayer and FileRecorder to Voice Engine
Because Voice Engine was the only user.

R=perkj@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/2037623002 .

Cr-Commit-Position: refs/heads/master@{#13261}
2016-06-22 21:47:53 +00:00
435f98bc90 Reland of Change ProcessThread's task type to be the one from TaskQueue. (patchset #1 id:1 of https://codereview.webrtc.org/2020783003/ )
Reason for revert:
Reland. It was a false alarm.

Original issue's description:
> Revert of Change ProcessThread's task type to be the one from TaskQueue. (patchset #3 id:80001 of https://codereview.webrtc.org/2016043003/ )
>
> Reason for revert:
> Downstream issues
>
> Original issue's description:
> > Change ProcessThread's task type to be the one from TaskQueue.
> > ProcessThread will eventually be replaced by TaskQueue, so this is the first little step.
> >
> > BUG=
> > R=magjed@webrtc.org
> >
> > Committed: https://crrev.com/400a276c8a0b299190ff17a81edd8780a26d63d3
> > Cr-Commit-Position: refs/heads/master@{#12952}
>
> TBR=magjed@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=
>
> Committed: https://crrev.com/641455176a1241dea6dda7071ba4162f41a0b5fc
> Cr-Commit-Position: refs/heads/master@{#12953}

TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review-Url: https://codereview.webrtc.org/2017333002
Cr-Commit-Position: refs/heads/master@{#12954}
2016-05-28 21:57:23 +00:00
641455176a Revert of Change ProcessThread's task type to be the one from TaskQueue. (patchset #3 id:80001 of https://codereview.webrtc.org/2016043003/ )
Reason for revert:
Downstream issues

Original issue's description:
> Change ProcessThread's task type to be the one from TaskQueue.
> ProcessThread will eventually be replaced by TaskQueue, so this is the first little step.
>
> BUG=
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/400a276c8a0b299190ff17a81edd8780a26d63d3
> Cr-Commit-Position: refs/heads/master@{#12952}

TBR=magjed@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=

Review-Url: https://codereview.webrtc.org/2020783003
Cr-Commit-Position: refs/heads/master@{#12953}
2016-05-28 18:00:38 +00:00
400a276c8a Change ProcessThread's task type to be the one from TaskQueue.
ProcessThread will eventually be replaced by TaskQueue, so this is the first little step.

BUG=
R=magjed@webrtc.org

Review URL: https://codereview.webrtc.org/2016043003 .

Cr-Commit-Position: refs/heads/master@{#12952}
2016-05-28 12:17:12 +00:00
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
ba35d05a49 Cleanup of iOS AudioDevice implementation
TBR=tkchin
BUG=webrtc:4789
TEST=modules_unittests --gtest_filter=AudioDeviceTest* and AppRTCDemo

Review URL: https://codereview.webrtc.org/1206783002 .

Cr-Commit-Position: refs/heads/master@{#9578}
2015-07-14 15:04:19 +00:00
c3f4dbc40b Remove rtp_rtcp/ dump functionality.
Removes RTP dumping from VideoEngine and VoiceEngine.

BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47179004

Cr-Commit-Position: refs/heads/master@{#9234}
2015-05-20 12:10:56 +00:00
b26198972c Adding support for OpenSL ES output in native WebRTC
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo

Summary:

- Removes dependency of the 'enable_android_opensl' compiler flag.
  Instead, OpenSL ES is always supported, and will enabled for devices that
  supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.

Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.

R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51759004

Cr-Commit-Position: refs/heads/master@{#9208}
2015-05-18 14:49:04 +00:00
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
749c60217d Moved gypi to avoid presubmit warning about '..' when touching the files.
R=kjellander@webrtc.org,mflodman@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39299004

Cr-Commit-Position: refs/heads/master@{#8503}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8503 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 11:50:44 +00:00