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15 Commits

Author SHA1 Message Date
kwiberg
84f6a3fc6b Move optional.h to webrtc/api/ 
We use Optional in our public API, so its header should be in
webrtc/api/.

BUG=webrtc:8205

Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
 2017-09-05 15:43:13 +00:00
Stefan Holmer
1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/. 
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
 2017-09-01 13:29:30 +00:00
minyue-webrtc
0e320ec5ba Wiring discard rate of audio FEC/RED packets up to StatsReport. 
BUG=webrtc:7903

Change-Id: I0325725be354ab89cfce1d3564936fe5ff93d303
Reviewed-on: https://chromium-review.googlesource.com/559339
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19560}
 2017-08-28 13:17:55 +00:00
Steve Anton
2dbc69fa64 Add stats totalSamplesReceived and concealedSamples 
Adds two new stats to RTCMediaStreamTrackStats:
* totalSamplesReceived is the total number of samples received on
      the audio channel and includes real and synthetic samples.
* concealedSamples is the total number of synthetic samples
      received on the audio channel used to conceal packet loss.

Bug: webrtc:8076
Change-Id: I36e9828525ec341490cf3310a972b56a8b443667
Reviewed-on: https://chromium-review.googlesource.com/615902
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19506}
 2017-08-25 00:50:42 +00:00
zstein
e76bd3aa43 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. 
BUG=webrtc:7982

Review-Url: https://codereview.webrtc.org/2964593002
Cr-Commit-Position: refs/heads/master@{#19027}
 2017-07-14 19:17:49 +00:00
Edward Lemur
c20978e581 Rename webrtc/base -> webrtc/rtc_base 
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
NOTRY=True
NOTREECHECKS=True
TBR=kwiberg@webrtc.org, kjellander@webrtc.org

Bug: webrtc:7634
Change-Id: I3cca0fbaa807b563c95979cccd6d1bec32055f36
Reviewed-on: https://chromium-review.googlesource.com/562156
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18919}
 2017-07-06 19:11:40 +00:00
Henrik Kjellander
a80c16a67c Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" 
This reverts commit c3771cc4d37f5573fe53b7c7cff295a4f0f9560f.
(breaks downstream internal project)

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2972463002 .
Cr-Commit-Position: refs/heads/master@{#18873}
 2017-07-01 14:48:18 +00:00
kjellander
c3771cc4d3 Update includes for webrtc/{base => rtc_base} rename (2/3) 
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.

Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
 2017-06-30 20:42:44 +00:00
hbos
8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) 
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
 2017-04-10 14:39:05 +00:00
olka
fbcc5cb386 Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) 
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
 2017-04-10 11:38:13 +00:00
zhihuang
292084c376 Added the GetSources() to the RtpReceiverInterface and implemented 
it for the AudioRtpReceiver.

This method returns a vector of RtpSource(both CSRC source and SSRC
source) which contains the ID of a source, the timestamp, the source
type (SSRC or CSRC) and the audio level.

The RtpSource objects are buffered and maintained by the
RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
the info of the contributing source will be pulled along the object
chain:
AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
AudioReceiveStream -> voe::Channel -> RtpRtcp module

Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource

BUG=chromium:703122
TBR=stefan@webrtc.org, danilchap@webrtc.org

Review-Url: https://codereview.webrtc.org/2770233003
Cr-Commit-Position: refs/heads/master@{#17591}
 2017-04-07 17:57:22 +00:00
solenberg
796b8f9d71 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream. 
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
 2017-03-02 01:02:23 +00:00
kwiberg
087bd34d23 Move AudioDecoder and related stuff to the api/ directory 
BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
 2017-02-10 16:15:44 +00:00
kwiberg
d32bf75721 Pass SdpAudioFormat through Channel, without converting to CodecInst 
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2516993002
Cr-Commit-Position: refs/heads/master@{#16165}
 2017-01-19 15:03:59 +00:00
ossu
f515ab8c3f Moved call.h and most of api/call/* into call/ 
BUG=webrtc:6716

Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
 2016-12-07 12:53:04 +00:00