6 Commits

Author SHA1 Message Date
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