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27 Commits

Author SHA1 Message Date
pbos
1ba8d39a9c Remove webrtc/stream.h and unutilized inheritance. 
Removes inheritance and a virtual call. Also removes a root header that
would have needed to be moved into a subdirectory otherwise to prevent
circular dependencies.

BUG=webrtc:4243
R=kjellander@webrtc.org, solenberg@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/1924793002
Cr-Commit-Position: refs/heads/master@{#12586}
 2016-05-02 03:18:36 +00:00
mflodman
3d7db263b9 Switch voice transport to use Call and Stream instead of VoENetwork. 
VoENetwork is kept for now, but is not really used anylonger.

webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.

BUG=webrtc:5079

TBR=tommi

Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
 2016-04-29 07:57:21 +00:00
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
kwiberg
fffa42b57e Replace scoped_ptr with unique_ptr in webrtc/audio/ 
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1706183002

Cr-Commit-Position: refs/heads/master@{#11723}
 2016-02-23 18:46:39 +00:00
Stefan Holmer
80e12072cf Move congestion controller to a separate module. 
This allows other projects to more easily depend on this.

The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h.

No functional changes in this CL.

R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1718473002 .

Cr-Commit-Position: refs/heads/master@{#11718}
 2016-02-23 12:30:51 +00:00
kwiberg
b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ 
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1702983002

Cr-Commit-Position: refs/heads/master@{#11658}
 2016-02-17 18:04:26 +00:00
stefan
ba4c0e45ff Add send-side BWE to WebRtcVoiceEngine under a finch experiment. 
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.

TBR=mflodman@webrtc.org
BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1604563002

Cr-Commit-Position: refs/heads/master@{#11487}
 2016-02-04 12:12:31 +00:00
stefan
bba9dec4d5 Use separate rtp module lists for send and receive in PacketRouter. 
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.

Also moves sending transport feedback to the pacer thread.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1628683002

Cr-Commit-Position: refs/heads/master@{#11443}
 2016-02-01 12:40:04 +00:00
stefan
3313ec901f Enable transport seq num extension on receive channel to suppress log warning. 
TBR=pbos@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1608563005

Cr-Commit-Position: refs/heads/master@{#11338}
 2016-01-21 14:32:48 +00:00
deadbeef
2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) 
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@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:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
 2016-01-13 20:00:29 +00:00
deadbeef
e591f9377f Storing raw audio sink for default audio track. 
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}
 2016-01-13 00:45:33 +00:00
Stefan Holmer
3842c5c7f7 Wire-up BWE feedback for audio receive streams. 
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
 2016-01-12 12:55:11 +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
358057b945 Use ChannelProxy for most calls on voe::Channel in Audio[Receive|Send]Stream. 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1482703002

Cr-Commit-Position: refs/heads/master@{#10828}
 2015-11-27 18:46:47 +00:00
solenberg
13725089ef Open backdoor in VoiceEngineImpl to get at the actual voe::Channel objects from an ID. 
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1459083007

Cr-Commit-Position: refs/heads/master@{#10788}
 2015-11-25 16:16:57 +00:00
solenberg
7add058439 Move some receive stream configuration into webrtc::AudioReceiveStream. 
Simplify creation of VoE channels and Call streams in WVoMC.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1454073002

Cr-Commit-Position: refs/heads/master@{#10731}
 2015-11-20 17:59:40 +00:00
solenberg
8b85de2ba1 Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail. 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1442483003

Cr-Commit-Position: refs/heads/master@{#10654}
 2015-11-16 17:48:12 +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
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include 
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
 2015-10-28 17:17:50 +00:00
solenberg
85a0496b8c Implement AudioSendStream::GetStats(). 
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1414743004

Cr-Commit-Position: refs/heads/master@{#10424}
 2015-10-27 10:35:30 +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
pbos
a2f30deea3 Log Call {audio, video} stream deletions. 
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1400333002

Cr-Commit-Position: refs/heads/master@{#10286}
 2015-10-15 12:22:21 +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