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13 Commits

Author SHA1 Message Date
ehmaldonado
e9cc686293 GN Templates: Move common_inherited_config to the template. 
Remove common_inherited_config from the targets and add it to the
template instead.

BUG=webrtc:6187
NOTRY=True

Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
 2016-09-05 13:10:23 +00:00
ehmaldonado
7a2ce0b738 GN Templates: Move common_config to the template. 
Remove common_config from the targets' config and add
it to the template instead.

BUG=webrtc:6187
NOTRY=True

Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
 2016-09-05 08:35:48 +00:00
ehmaldonado
38a2132b02 GN: Introduce templates. 
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.

These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target

Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.

BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True

Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
 2016-09-02 11:10:41 +00:00
kjellander
a69d973267 Move webrtc/audio_*.h to webrtc/api/call 
BUG=webrtc:5878
NOTRY=True

Review-Url: https://codereview.webrtc.org/2059703002
Cr-Commit-Position: refs/heads/master@{#13996}
 2016-08-31 14:33:14 +00:00
Peter Boström
0208322ee3 GN: Add video_engine_tests 
Adds separate source_sets for the video_engine_tests subtargets inside
audio, call and video and merges them together into video_engine_tests.

BUG=webrtc:5949
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/2064523002 .

Cr-Commit-Position: refs/heads/master@{#13127}
 2016-06-14 10:53:09 +00:00
kjellander@webrtc.org
50772f1e16 GN: Update audio_sink.h location 
This should have been a part of
https://codereview.webrtc.org/1740873003/
but wasn't discovered since we cannot have --check turned
on for GN yet.

BUG=webrtc:5589
TBR=solenberg@webrtc.org, henrika@webrtc.org,

Review URL: https://codereview.webrtc.org/1744123002 .

Cr-Commit-Position: refs/heads/master@{#11803}
 2016-02-29 05:49:54 +00:00
Tommi
f888bb58da Support for unmixed remote audio into tracks. 
BUG=chromium:121673
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1505253004 .

Cr-Commit-Position: refs/heads/master@{#10995}
 2015-12-12 00:37:14 +00:00
solenberg
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats(). 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
 2015-11-06 23:34:58 +00:00
Fredrik Solenberg
4f4ec0a927 Re-Land: Implement AudioReceiveStream::GetStats(). 
R=tommi@webrtc.org
BUG=webrtc:4690

Committed: a457752f4a

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10369}
 2015-10-22 08:49:39 +00:00
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ ) 
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
 2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats(). 
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
 2015-10-20 13:01:55 +00:00
solenberg
c7a8b08a7c Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. 
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1397123003

Cr-Commit-Position: refs/heads/master@{#10307}
 2015-10-16 21:35:11 +00:00
Peter Boström
5c389d3e09 Split webrtc/video into webrtc/{audio,call,video}. 
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.

BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1227923005 .

Cr-Commit-Position: refs/heads/master@{#10073}
 2015-09-25 11:58:39 +00:00