10 Commits

Author SHA1 Message Date
kjellander@webrtc.org
7ffeab525c Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
This is a reland of https://codereview.webrtc.org/1737593002/ minus
the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and
webrtc/common.gyp that breaks GN in Chromium since it's using
the --check flag (which we should support).

BUG=webrtc:4243, webrtc:5589
TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully.
TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1740873003 .

Cr-Commit-Position: refs/heads/master@{#11794}
2016-02-26 21:46:22 +00:00
kjellander
7324eb9e62 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
Reason for revert:
Breaks GN in chromium.

Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}

TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589

Review URL: https://codereview.webrtc.org/1739783002

Cr-Commit-Position: refs/heads/master@{#11769}
2016-02-25 16:37:02 +00:00
kjellander@webrtc.org
99b345c4e5 Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
depending on voice engine, resulting in a cyclic dependency (which we
don't detect since we have that check turned off, see webrtc:4243).

BUG=webrtc:4243, webrtc:5589
R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1737593002 .

Cr-Commit-Position: refs/heads/master@{#11766}
2016-02-25 14:12:48 +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