16 Commits

Author SHA1 Message Date
aleloi
aed581a4f3 Made AudioReceiveStream a mixer participant.
Methods to facilitate this are added to ChannelProxy and voe::Channel.

BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2378143004
Cr-Commit-Position: refs/heads/master@{#14707}
2016-10-20 13:32:47 +00:00
kjellander
e40a7ee007 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
These suppressions are causing GN errors when Chromium targets are depending
directly on WebRTC targets (needed for https://codereview.chromium.org/2413103004)

BUG=webrtc:4256
NOTRY=True

Review-Url: https://codereview.webrtc.org/2408133008
Cr-Commit-Position: refs/heads/master@{#14644}
2016-10-17 06:56:20 +00:00
kjellander
b62dbbe985 GN: Change rtc_source_set targets --> rtc_static_library
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).

After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()

See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.

NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.

BUG=webrtc:6410, chromium:630755

Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
2016-09-23 07:38:58 +00:00
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