| 566ef247b9779f6c9d0e7ec9eea6b037f4682c53 |
|
07-Nov-2015 |
solenberg <solenberg@webrtc.org> |
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}
/external/webrtc/webrtc/call.h
|
| c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/call.h
|
| cf18b34cf3bbb1cc2984f8ae3a1c5cebf92b7007 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Align new VoE API with design.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1376153003
Cr-Commit-Position: refs/heads/master@{#10136}
/external/webrtc/webrtc/call.h
|
| 68786d20400f1f3744ad83549325665c18ea9e5b |
|
08-Sep-2015 |
stefan <stefan@webrtc.org> |
Wire up PacketTime to ReceiveStreams.
BUG=webrtc:4758
Review URL: https://codereview.webrtc.org/1333483002
Cr-Commit-Position: refs/heads/master@{#9892}
/external/webrtc/webrtc/call.h
|
| e5269747595864eedd604f153df5d7bcbe1b475a |
|
08-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementation slightly simpler. See discussion in: https://codereview.webrtc.org/1269863005/
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1325263002
Cr-Commit-Position: refs/heads/master@{#9891}
/external/webrtc/webrtc/call.h
|
| 4fbae2b79134572135d9d5fe35a7d1ccdeea3a4d |
|
28-Aug-2015 |
solenberg <solenberg@webrtc.org> |
Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1273363005
Cr-Commit-Position: refs/heads/master@{#9807}
/external/webrtc/webrtc/call.h
|
| cd6702282a49448adda470934f4bd9e6181cab22 |
|
16-Jul-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
Define Stream base classes
BUG=webrtc:4690
Defined classes Stream, SendStream and ReceiveStream. Inherited existing stream classes from either SendStream or ReceiveStream.
This is a step towards having a Transport associated with streams instead of a Call. It will allow a lot of code in the Call to be media type agnostic.
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1226123005 .
Cr-Commit-Position: refs/heads/master@{#9591}
/external/webrtc/webrtc/call.h
|
| 78fb3b3f8f8f46e3b5b62c94177713dc05f53947 |
|
11-Jun-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
C++11 in-class member initialization in Call configs.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1166263004.
Cr-Commit-Position: refs/heads/master@{#9416}
/external/webrtc/webrtc/call.h
|
| 04f4931ef06273c2873e7816ed1f568d445117b8 |
|
08-Jun-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
VoE2 API draft
BUG=4690
R=jmarusic@webrtc.org, kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50029004
Cr-Commit-Position: refs/heads/master@{#9392}
/external/webrtc/webrtc/call.h
|
| 23fba1ffa0079f70744a83bcf4e85501dc226013 |
|
29-Apr-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Add AudioReceiveStream to Call API.
BUG=4574
R=kwiberg@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51749004
Cr-Commit-Position: refs/heads/master@{#9114}
/external/webrtc/webrtc/call.h
|
| 76c53d36bc455fe89ca1f860d5171633198fe907 |
|
09-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViE interface usage from VideoReceiveStream.
References channels and underlying objects directly instead of using
interfaces referenced with channel id. Channel creation is still done as
before for now.
BUG=1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46849004
Cr-Commit-Position: refs/heads/master@{#8958}
/external/webrtc/webrtc/call.h
|
| e59041672283a28bde0b043c0c2bc198272f82e1 |
|
26-Mar-2015 |
Stefan Holmer <holmer@google.com> |
Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
/external/webrtc/webrtc/call.h
|
| 16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 2b19f0631233488e891d9db0d170b637dc8fc464 |
|
11-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up RTT statistics to webrtc::Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/32249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 008731868a09e2fe01da53733a612dc24761f791 |
|
25-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement settable min/start/max bitrates in Call.
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/26199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 0bae1fab4adb9bb8164e53142bf419049eafec38 |
|
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up bandwidth stats to the new API and webrtcvideoengine2.
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| a73a678e25bfec7d89a1a241419897ee8a64f7b0 |
|
14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove -1 from Call::Config::start_bitrate_bps.
Instead initialize it to a good default value. The code does the same,
but we don't have to check explicitly for -1.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/23989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7445 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 42684be21b255e2b07eb154e6a2807fa2226167e |
|
03-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up CPU adaptation in WebRtcVideoEngine2.
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.
BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| bbe0a8517d7f9da7aa779bff77cdbb70df358437 |
|
19-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Config struct for VideoEncoder.
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 26c0c41a06d77af54df547169d952a21319dea8c |
|
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Network up/down signaling in Call.
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| bd249bc711b3c9efd142eb8de3df489282fe693e |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove GetDefaultConfigs() from Call.
Defaults for configs are instead placed in the Config constructors.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| eb16b811fb5c21954f9e4a6f4c64e297e9fd65b9 |
|
16-Jun-2014 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implements start bitrate for new video API.
Added a new rampup test.
BUG=2879
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 6ae48c660934784b4df56ab1ac99402ce3745e9f |
|
06-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make VideoSendStream/VideoReceiveStream configs const.
Benefits of this is that the send config previously had unclear locking
requirements, a lock was used to lock parts parts of it while
reconfiguring the VideoEncoder. Primary work was splitting out video
streams from config as well as encoder_settings as these change on
ReconfigureVideoEncoder. Now threading requirements for both member
configs are clear (as they are read-only), and encoder_settings doesn't
stay in the config as a stale pointer.
CreateVideoSendStream now takes video streams separately as well as the
encoder_settings pointer, analogous to ReconfigureVideoEncoder.
This change required changing so that pacing is silently enabled when
using suspend_below_min_bitrate rather than silently setting it.
R=henrik.lundin@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
BUG=3260
Review URL: https://webrtc-codereview.appspot.com/20409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6349 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| caba2d2a370cb6b5e67c881ecfa57fdac7411de8 |
|
14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add DeliveryStatus enum to DeliverPacket().
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 5ca6a5387e5817d7753a35382d6cf6234b3f4acc |
|
24-Apr-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove TraceCallback use from Call.
Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.
R=mflodman@webrtc.org
BUG=3250,3157
Review URL: https://webrtc-codereview.appspot.com/12329006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| f577ae9eac9822380ea6f0fb953cf383d0ec5374 |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove internal codecs from VideoSendStream.
Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.
For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.
Removes Call::GetVideoCodecs().
Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.
BUG=2854,2992
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| bdc5ed2e7d081de6597f0f993d54489eb3abd496 |
|
31-Jan-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add configuration for cpu overuse detection to video send stream.
BUG=2422
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 39669c5c8fd71c99b5f8e2015f09d178d866ff47 |
|
07-Jan-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove outdated DestroyVideoSendStream comment.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5345 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| b429e516a98d2dee0c57d3263f6d21633939b564 |
|
18-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
cpplint cleaning new API and its implementation files.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 7e9315b42ebe8f7df860030af93618de81326503 |
|
04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for sending redundant payloads over RTX.
TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 2c46f8d854c1fc3e10f8151ee5109923287aee8b |
|
21-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename DestroyStream methods to include Video.
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 5a63655ab0de18bd2fa376ba4774eab3f3bc9fb2 |
|
20-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename Call::Create{Receive,Send}Stream().
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 6488761f2e6ce7b977bbc14bc7b91933527d633a |
|
14-Nov-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement VideoSendStream::SetCodec().
Removing assertion that SSRC count should be the same as the number of
streams in the codec. It makes sense that you don't always use the same
number of streams under one call. Dropping resolution due to CPU overuse
for instance can require less streams, but the SSRCs should stay
allocated so that operations can resume when not overusing any more.
This change also means we can get rid of the ugly SendStreamState whose
content wasn't defined. Instead we use SetCodec to change resolution
etc. on the fly. Should something else have to be replaced on the fly
then that functionality simply has to be implemented.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3499005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5123 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|
| 16e03b7bd8b88ba569987e20a7f29061f91a3d0d |
|
28-Oct-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/call.h
|