21 Commits

Author SHA1 Message Date
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
mflodman
0c478b3d75 Rename ChannelGroup to CongestionController and move to webrtc/call/.
BUG=webrtc:5079
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1419803002 .

Cr-Commit-Position: refs/heads/master@{#10358}
2015-10-21 13:52:33 +00:00
mflodman
e37870297f ChannelGroup cleanup.
Move CallStats to Call, EncoderStateFeedback to VideoSendStream and
remove last ViEChannel dependency from ChannelGroup.

BUG=webrtc:5079
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1418613002 .

Cr-Commit-Position: refs/heads/master@{#10355}
2015-10-21 11:24:37 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
2015-10-21 06:00:57 +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
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
terelius
1adce14c87 Old config events are no longer removed from RtcEventLog.
Time to keep old events in buffer is now changeable at runtime.
Added unit test for removing old events from buffer.

Review URL: https://codereview.webrtc.org/1303713002

Cr-Commit-Position: refs/heads/master@{#10302}
2015-10-16 15:51:15 +00:00
ivoc
112a3d81db Added functions on libjingle API to start and stop the recording of an RtcEventLog.
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1374253002

Cr-Commit-Position: refs/heads/master@{#10297}
2015-10-16 09:22:23 +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
Ivo Creusen
301aaed813 Update to the RtcEventLog protobuf to remove the DebugEvent message.
This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.

This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.

BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/1348113003 .

Cr-Commit-Position: refs/heads/master@{#10221}
2015-10-08 16:07:53 +00:00
stefan
1d8a506405 Add a PacketOptions struct to webrtc::Transport.
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.

BUG=4173

Review URL: https://codereview.webrtc.org/1376673004

Cr-Commit-Position: refs/heads/master@{#10144}
2015-10-02 10:39:40 +00:00
pbos
da903eaabb Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1373903003

Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 09:37:18 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces.
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +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