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60 Commits

Author SHA1 Message Date
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
mflodman
0dbf0090a9 Remove the video channel id completely. 
BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1412143002

Cr-Commit-Position: refs/heads/master@{#10324}
 2015-10-19 15:12:19 +00:00
solenberg
5a28939392 Added thread checker to webrtc::Call. 
BUG=

Review URL: https://codereview.webrtc.org/1403353003

Cr-Commit-Position: refs/heads/master@{#10321}
 2015-10-19 10:39:27 +00:00
mflodman
a20de2030f Move ownership of receive ViEChannel to VideoReceiveStream. 
This CL changes as little as possible and I'll follow up later with
ownership of the other members in ChannelGroup.

The next step is to remove the id used for channels.

BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1411723002

Cr-Commit-Position: refs/heads/master@{#10318}
 2015-10-19 05:08:29 +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
stefan
c1aeaf0dc3 Wire up packet_id / send time callbacks to webrtc via libjingle. 
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1363573002

Cr-Commit-Position: refs/heads/master@{#10289}
 2015-10-15 14:26:17 +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
stefan
457a61db61 Pause/resume pacer from Call instead of via SendStreams. 
BUG=webrtc:5073

Review URL: https://codereview.webrtc.org/1398443007

Cr-Commit-Position: refs/heads/master@{#10271}
 2015-10-14 10:13:04 +00:00
stefan
4fbd145dce Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side. 
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.

BUG=webrtc:4836

Review URL: https://codereview.webrtc.org/1368943002

Cr-Commit-Position: refs/heads/master@{#10087}
 2015-09-28 10:57:23 +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