| b7d9a97ce41022e984348efb5f28bf6dd6c6b779 |
|
18-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Expose codec implementation names in stats.
Used to distinguish between software/hardware encoders/decoders and
other implementation differences. Useful for tracking quality
regressions related to specific implementations.
BUG=webrtc:4897
R=hta@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1406903002 .
Cr-Commit-Position: refs/heads/master@{#11084}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 17821db19702aca15d0d93cb60515ca70823fad7 |
|
14-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Wire up bandwidth limitation info to GetStats and adapt_reason.
The input resolution (output from video_adapter) can be further scaled down or higher video layer(s) can be dropped due to bitrate constraints.
BUG=webrtc:4112
Review URL: https://codereview.webrtc.org/1502173002
Cr-Commit-Position: refs/heads/master@{#11006}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 822bdf978435b8eba9343ea96e9a9bc54b9c7df0 |
|
11-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Remove cricket::VideoEncoderConfig.
BUG=webrtc:5332
R=noahric@chromium.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1512853007 .
Cr-Commit-Position: refs/heads/master@{#10991}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1387149ad1669365ac05278bf779a407bec08a4e |
|
09-Dec-2015 |
deadbeef <deadbeef@webrtc.org> |
Adding reduced size RTCP configuration down to the video stream level.
Still waiting to turn on negotiation (in mediasession.cc)
until we verify it's working as expected.
BUG=webrtc:4868
Review URL: https://codereview.webrtc.org/1418123003
Cr-Commit-Position: refs/heads/master@{#10958}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9f45a45a628100d973111b3aac66dede57454b6a |
|
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Add tracing to upper-level WebRTC calls.
Adds tracing to WebRtcSession and corresponding BaseChannel calls to see
where time is spent better.
BUG=webrtc:5167
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1505023003 .
Cr-Commit-Position: refs/heads/master@{#10934}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9d69c3f4d99240c27d997c37950b561605d403bd |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Return a copy of the supported RTP header extensions instead of a reference.
This also renames the method to better reflect what it does.
BUG=webrtc:5187
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1486123002 .
Cr-Commit-Position: refs/heads/master@{#10910}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9e1b992f74470aecfeb216e26b455982ddc4a6d5 |
|
04-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Clear old decoders after recreating the receiver.
Prevents UAF when switching decoder capabilities and the
previously-supported decoder is currently being received on.
BUG=chromium:565967
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1490233010 .
Cr-Commit-Position: refs/heads/master@{#10898}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 7e4e01a4413fa98644b94ab9d8a9dccc664f39f2 |
|
02-Dec-2015 |
solenberg <solenberg@webrtc.org> |
Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
Rework filtering functionality to be reused for both Audio+Video.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1481963002
Cr-Commit-Position: refs/heads/master@{#10869}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 444682acf9804c5fcbddaded9e900ba3cc6921fc |
|
25-Nov-2015 |
qiangchen <qiangchen@chromium.org> |
Remove frame time scheduing in IncomingVideoStream
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 43edf0ffb91a50e2efa01c7befe4d188a7e30ea2 |
|
21-Nov-2015 |
stefan <stefan@webrtc.org> |
Require negotiation to send transport cc feedback over RTCP.
BUG=4312
Review URL: https://codereview.webrtc.org/1452883002
Cr-Commit-Position: refs/heads/master@{#10735}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| bd13838ccc87f94d1e951bcf780979622b020359 |
|
21-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1457653003
Cr-Commit-Position: refs/heads/master@{#10734}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 2aff615bd7c7c24a6e7a35163112f169ff4f9246 |
|
18-Nov-2015 |
Peter Boström <pbos@webrtc.org> |
Remove spammy logging of RTCP delivery failures.
Since BundleFilter doesn't filter RTCP anymore we can have incoming
RTCPs for audio delivered to video, that delivery will fail when there
are no video receivers causing the log to be spammed.
BUG=webrtc:5223
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1458853002 .
Cr-Commit-Position: refs/heads/master@{#10687}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| be57983f4bd875c39a229bab5112b32dad004057 |
|
10-Nov-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Rename Maybe to Optional
And add examples of good and bad usage to the documentation.
R=aluebs@webrtc.org, henrik.lundin@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1432553007 .
Cr-Commit-Position: refs/heads/master@{#10588}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3ed348707e93980fd74246f7a1dfab011f841087 |
|
10-Nov-2015 |
asapersson <asapersson@webrtc.org> |
Remove field trial check for VP9.
VP9 is put as second codec in supported codec list.
BUG=chromium:500602
Review URL: https://codereview.webrtc.org/1432673002
Cr-Commit-Position: refs/heads/master@{#10577}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 102c6a61bc0b42dc0956d013530fc0213b7e881b |
|
30-Oct-2015 |
kwiberg <kwiberg@webrtc.org> |
Replace rtc::cricket::Settable with rtc::Maybe
The former is very similar to the latter, but less general (mostly in
naming).
This CL, which is the first to use Maybe at scale, also removes the implicit conversion from T to Maybe<T>, since it was agreed that the increased verbosity increased legibility.
Review URL: https://codereview.webrtc.org/1430433004
Cr-Commit-Position: refs/heads/master@{#10461}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| bbaf3633c54e3d49aa4c762b8eaa34e09de01c45 |
|
29-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Filter overlapping RTP header extensions.
This removes unnecessary RTP header extension overhead since only one of
these extensions is used at a time.
BUG=webrtc:4254
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1429753003 .
Cr-Commit-Position: refs/heads/master@{#10455}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include
BUG=webrtc:5095
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1413333002 .
Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 4cba4eba596706f2238d14f96f4e181f47e5034c |
|
26-Oct-2015 |
pbos <pbos@webrtc.org> |
Disable denoising for VP9 by default.
BUG=webrtc:5108
R=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1418133012
Cr-Commit-Position: refs/heads/master@{#10413}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| dfa2815b4f606a58ede5c0214e08a1d5d26d3639 |
|
21-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Update receive report SSRCs on RemoveSendStream.
Prevents RTCP receiver reports, including PLIs with an old
receiver-report SSRC, from being dropped from the remote sender's
BundleFilter due to no longer being in use.
BUG=chromium:523928, webrtc:4883
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1404363003 .
Cr-Commit-Position: refs/heads/master@{#10359}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e4f96501fc5b3e6de0d1ccd262372afcda1f5b4f |
|
21-Oct-2015 |
tommi <tommi@webrtc.org> |
Remove system_wrappers/interface/trace_event.h
BUG=
Review URL: https://codereview.webrtc.org/1417773002
Cr-Commit-Position: refs/heads/master@{#10346}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| be16f79818d7c21b747189b3e86d8d98add3e6b1 |
|
16-Oct-2015 |
pbos <pbos@webrtc.org> |
Remove simulcast bitrate modes.
Instead always use the SBM_VERY_HIGH setting.
BUG=webrtc:4885
R=hta@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1407693005
Cr-Commit-Position: refs/heads/master@{#10305}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d97ec30ce4f22ba2d88314d67ff44458144a5096 |
|
07-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1386653002
Cr-Commit-Position: refs/heads/master@{#10194}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1d8a506405734d0cef9653704b036ca4f1388960 |
|
02-Oct-2015 |
stefan <stefan@webrtc.org> |
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}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b09b660c53ff2c499d149e05e5c435f5057273fc |
|
01-Oct-2015 |
magjed <magjed@webrtc.org> |
Remove cricket::VideoFrame::Set/GetElapsedTime()
This CL is a baby step towards consolidating the timestamps in cricket::VideoFrame and webrtc::VideoFrame, so that we can unify the frame classes in the future.
The elapsed time functionality is not really used. If a video sink wants to know the elapsed time since the first frame they can store the first timestamp themselves and calculate the time delta to later frames. This is already done in all video sinks that need the elapsed time. Having redundant timestamps in the frame classes is confusing and error prone.
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1324263004
Cr-Commit-Position: refs/heads/master@{#10131}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| dfc8f4ff8731390828884a0a91b99e51f2950275 |
|
01-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1378513003
Cr-Commit-Position: refs/heads/master@{#10130}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 34fbfff068bf46d27812fb8fd531aea889a5feaf |
|
24-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoMediaChannel::SetRender().
Was a no-op in current implementation.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1334793003 .
Cr-Commit-Position: refs/heads/master@{#10059}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 61e933eac7673feb2f8663c3e71e503b714b350f |
|
24-Sep-2015 |
solenberg <solenberg@webrtc.org> |
Remove ChannelManager::GetCapabilities()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1364083002
Cr-Commit-Position: refs/heads/master@{#10045}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b071a19019a0a2173cc139c960d6ef6946a1c581 |
|
17-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1327933002 .
Cr-Commit-Position: refs/heads/master@{#9973}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 709ed67c38d0a942f3bf3e68e337cc27a27bc353 |
|
15-Sep-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).
BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1269863005 .
Cr-Commit-Position: refs/heads/master@{#9939}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1cb121dea478a4bb4f88e76cf92719e2853543cf |
|
14-Sep-2015 |
pbos <pbos@webrtc.org> |
Reset frame timestamp epoch for new capturers.
Incoming frames usually have an epoch of time since the capturer was
created or similar, not any fixed-time epoch. As such, setting a new
capturer resulted in delivering frames with older timestamps which
caused these frames to be dropped before encoding.
BUG=webrtc:4994
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1345473002
Cr-Commit-Position: refs/heads/master@{#9934}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1dd98f321920c1442dd5b3f791ea0fca133c2756 |
|
10-Sep-2015 |
solenberg <solenberg@webrtc.org> |
- Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel)
- Rename VideoChannel::MuteStream() -> SetVideoSend() (incl. media channel)
- Collapse NnChannel::SetChannelOptions() into the above.
- Collapse VoiceChannel::SetLocalRenderer into SetAudioSend().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1311533009
Cr-Commit-Position: refs/heads/master@{#9915}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 2feafdb742226f57588d9c95bc25b2202166688f |
|
09-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Enable automatic resizing for RTX-enabled senders.
These were accidentally disabled due to checking ssrcs_.size() (which
includes RTX SSRCs) instead of rtp.ssrcs.size() to determine whether a
stream is simulcast or not.
BUG=webrtc:4965
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1318193003 .
Cr-Commit-Position: refs/heads/master@{#9907}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1c7d48d431e098ba42fa6bd9f1cfe69a703edee5 |
|
08-Sep-2015 |
Åsa Persson <asapersson@webrtc.org> |
Let max default bitrate depend on resolution when configuring one video stream (was previously always 2Mbps).
Is now set to:
<= 320x240: 600kbps
<= 640x480: 1.7Mbps
<= 960x540: 2Mbps
> 960x540: 2.5Mbps
For QVGA and VGA, the qp was around 10 at the selected thresholds when running some tests. The change in qp declined above the selected bitrates.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1297373003 .
Cr-Commit-Position: refs/heads/master@{#9878}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 47d78cc8ad54baabc9042c2b848ae3afd9b80d2e |
|
04-Sep-2015 |
sophiechang <sophiechang@chromium.org> |
Pass the encoder's internal source property through to video_sender to request a keyframe from the external encoder
BUG=
Review URL: https://codereview.webrtc.org/1263663005
Cr-Commit-Position: refs/heads/master@{#9853}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f42376c60111edba6f29060bf3dd79e75d8dbb97 |
|
28-Aug-2015 |
pbos <pbos@webrtc.org> |
Wire up currently-received video codec to stats.
BUG=webrtc:1844, webrtc:4808
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1315413002
Cr-Commit-Position: refs/heads/master@{#9810}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| fdac516510a2bd5d57b0786fbd49e2a6b9aeed2f |
|
27-Aug-2015 |
noahric <noahric@chromium.org> |
Disallow simulcast for H.264.
BUG=
Review URL: https://codereview.webrtc.org/1291673006
Cr-Commit-Position: refs/heads/master@{#9795}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 85ad62b87760213a1a453051d833c8c40e82d9bd |
|
26-Aug-2015 |
Noah Richards <noahric@chromium.org> |
Remove per-frame captured frame logging.
It's a little too verbose :)
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1302173004 .
Cr-Commit-Position: refs/heads/master@{#9786}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 874ca3af5b163e1b3fd8802171e44ee252557842 |
|
21-Aug-2015 |
deadbeef <deadbeef@webrtc.org> |
Don't do reconfiguration if recv codec order/preference changes
Adding 'ReceiveCodecsHaveChanged' method that will determine if codecs
HAVE changed, irrespective of order and preference.
Review URL: https://codereview.webrtc.org/1291763003
Cr-Commit-Position: refs/heads/master@{#9748}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c2ee2c86f905991a8cd05ee1f35bea105b41e4e0 |
|
08-Aug-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1229283003 .
Cr-Commit-Position: refs/heads/master@{#9690}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c27d89fdc6b33846ff06e8ca8bd03119d05c6530 |
|
16-Jul-2015 |
qiangchen <qiangchen@chromium.org> |
Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to output frame.
Essentially we are carrying over the capture timestamp to the encoded frame sent out, so the frame lengths will contain no noise.
Review URL: https://codereview.webrtc.org/1225153002
Cr-Commit-Position: refs/heads/master@{#9597}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| bd3842808996dbb85007242214352f1e6ebd3d17 |
|
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Don't use result of "field_trial::FindFullName" as string reference.
"field_trial::FindFullName" can return "std::string()" which should not
be referenced by the caller.
Review URL: https://codereview.webrtc.org/1238943003
Cr-Commit-Position: refs/heads/master@{#9594}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 083b73fb95755b78cb0b9cbe67752b7e7b7eb263 |
|
16-Jul-2015 |
jbauch <jbauch@webrtc.org> |
Use std::string references instead of copying contents.
This CL improves the memory footprint a bit by using string references
instead of creating a copy.
Review URL: https://codereview.webrtc.org/1241973002
Cr-Commit-Position: refs/heads/master@{#9592}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8fc7fa798f7a36955f1b933980401afad2aff592 |
|
15-Jul-2015 |
pbos <pbos@webrtc.org> |
Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.
BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1181653002
Cr-Commit-Position: refs/heads/master@{#9586}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d10a68e7974a29b26d6c926e6f137255f3c173be |
|
10-Jul-2015 |
noahric <noahric@chromium.org> |
Don't create unsignalled receive streams for RTX, RED RTX, and ULPFEC packets.
BUG=webrtc:4389
Review URL: https://codereview.webrtc.org/1226093002
Cr-Commit-Position: refs/heads/master@{#9566}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 71f6f4405c1c5f60097f8d10841378088e78e8b9 |
|
29-Jun-2015 |
Zeke Chin <tkchin@webrtc.org> |
iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.
BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1187573004.
Cr-Commit-Position: refs/heads/master@{#9515}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 4765070b8d6f024509c717c04d9b708750666927 |
|
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
Rename I420VideoFrame to VideoFrame.
This is a mechanical change since it affects so many
files.
I420VideoFrame -> VideoFrame
and reformatted.
Rationale: in the next CL I420VideoFrame will
get an indication of Pixel Format (I420 for
starters) and of storage type: usually
UNOWNED, could be SHMEM, and in the near
future will be possibly TEXTURE. See
https://codereview.chromium.org/1154153003
for the change that happened in Cr.
BUG=4730, chromium:440843
R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52629004
Cr-Commit-Position: refs/heads/master@{#9339}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 259bd2034c3d3ee7f2dc4d481e9bf896a3a4d6ef |
|
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Report ssrc_groups in GetStats().
This was already available in the stats struct, just not filled in.
BUG=4720
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47329004
Cr-Commit-Position: refs/heads/master@{#9308}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3b187b9c0c4f404c31523e74909ee3b75a83846e |
|
28-May-2015 |
Henrik Boström <hbos@webrtc.org> |
Removed unnecessary includes of webrtcvideocapturer.h
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57469004
Cr-Commit-Position: refs/heads/master@{#9305}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3548dd21542c7b3f2c4680c6a6d86b0d719bd008 |
|
22-May-2015 |
Peter Boström <pbos@webrtc.org> |
Set local SSRCs on receivers added before senders.
Addresses bug where a receiver would report SSRC 1 even though the
endpoint has sending streams.
BUG=4678
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51099004
Cr-Commit-Position: refs/heads/master@{#9262}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9a416bd14ee225d8f1a1ada627a1dd7daf275032 |
|
22-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Get rid of unnecessary Terminate() method and worker_thread_ from WebRtcVideoEngine2
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51879004
Cr-Commit-Position: refs/heads/master@{#9258}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 4d71edef45afa38b3f68b6af0519ac0f21d327df |
|
19-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software encoding.
Permits falling back to software encoding for unsupported resolutions.
BUG=chromium:475116, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46279004
Cr-Commit-Position: refs/heads/master@{#9227}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 7252a2ba8035c4128917a9558a3e34fc9dbe7c44 |
|
18-May-2015 |
Peter Boström <pbos@webrtc.org> |
Add HW fallback option to software decoding.
Permits falling back to software decoding for unsupported resolutions in
bitstreams.
BUG=4625, chromium:487934
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46269004
Cr-Commit-Position: refs/heads/master@{#9209}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 67c9df79828991c5aab96b9253ae4e7ba7ed508e |
|
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base NACK on send codecs.
Addressing discrepancy where NACK used to be set from send codecs in
WebRtcVideoEngine(1), and before this change, from recv codecs in
WebRtcVideoEngine2. This should address that NACK might be sent even if
the remote side does not support it.
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53409004
Cr-Commit-Position: refs/heads/master@{#9171}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 126c03ea02d8a99bfa3d1e6d6fe04516183d31af |
|
11-May-2015 |
Peter Boström <pbos@webrtc.org> |
Base decision to send REMB on send codecs.
Fixes bug where Chromium would send REMB even though the remote party
doesn't announce support for it (because it was based on local codec
settings instead of remote ones).
BUG=4626
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54389004
Cr-Commit-Position: refs/heads/master@{#9170}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d3ddc1b69e9cdfd7c6d38ab02b8d8ab891d30fd1 |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
BUG=
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49929004
Cr-Commit-Position: refs/heads/master@{#9156}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 4b60c73e74d62beff484b7f54d8f3267cb66274f |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49269004
Cr-Commit-Position: refs/heads/master@{#9150}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 81ea54eaac82b36b7208a02fd37a469d7d0bd9d0 |
|
07-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove WebRtcVideoEngine.
Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until
build files in Chromium have been modified.
BUG=1695,4566
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48339004
Cr-Commit-Position: refs/heads/master@{#9148}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e433c0ef31297d78336d99cc18cf063b1a486cf2 |
|
01-May-2015 |
Alex Glaznev <glaznev@google.com> |
Restore back verbosity logging for camera captured frame.
Helps to debug camera freezes.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46179004
Cr-Commit-Position: refs/heads/master@{#9127}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| efbde3775b5eed8015d7e2e86ddcea3e6033d321 |
|
29-Apr-2015 |
Erik Språng <sprang@webrtc.org> |
Don't use CPU adaptation for screen content in the new API.
BUG=4605
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48309004
Cr-Commit-Position: refs/heads/master@{#9116}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| faa6d076b7d021467cc05e7c293940773524cad8 |
|
28-Apr-2015 |
Alex Glaznev <glaznev@google.com> |
Remove a few verbose log messages from webrtcvideoengine2.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49189004
Cr-Commit-Position: refs/heads/master@{#9105}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 143cec1cc68b9ba44f3ef4467f1422704f2395f0 |
|
28-Apr-2015 |
Erik Språng <sprang@google.com> |
Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.
BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46069004
Cr-Commit-Position: refs/heads/master@{#9093}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ee0b00e8a9cc2d8f4578912a389dee92ac020ee9 |
|
22-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent recv-stream reconfig on identical codecs.
Receive streams seem to be reconfigured with identical codecs when
another stream is removed. Preventing this reconfiguration makes sure
that existing streams don't report stats during teardown when the stream
is still supposed to be running.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44249004
Cr-Commit-Position: refs/heads/master@{#9059}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 393347ff988708df5037ddcd181fe204bd1ab37e |
|
22-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Report receive-side packet loss.
BUG=4558
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48219004
Cr-Commit-Position: refs/heads/master@{#9054}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 7c027b64ae53a29bc528b4241cc540694c239304 |
|
22-Apr-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46999004
Cr-Commit-Position: refs/heads/master@{#9053}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e62202fedf57b74cc263246c0586ee353978caf8 |
|
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.
Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.
BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36889004
Cr-Commit-Position: refs/heads/master@{#9040}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3c3f6460646183914629e5dab8ae5fcede4f0e80 |
|
15-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Prevent null-stream reconfigs on RTP extensions.
If a codec fails to set (e.g. there's no codec configured), this
prevents a stream reconfigure with an invalid config. Reconfiguring a
stream without correct codec settings causes a CHECK failure.
BUG=chromium:475116
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44219004
Cr-Commit-Position: refs/heads/master@{#9007}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e432800aeb6b695bda14acf2d60c0200803b5218 |
|
14-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Enable CPU adaptation by default.
WebRtcVideoEngine2 doesn't support CPU-monitor-based adaptation and as
such requires encoder-time-based CPU adaptation to perform any
adaptation at all.
BUG=4536
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49679004
Cr-Commit-Position: refs/heads/master@{#9001}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 77f0e3f7b6a0664661dc295eb235c543b8091554 |
|
13-Apr-2015 |
Peter Thatcher <pthatcher@chromium.org> |
Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48039004
Cr-Commit-Position: refs/heads/master@{#8990}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e7b221f4760af10e29cb4c501e758cc3518f628b |
|
13-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove deadlock in WebRtcVideoEngine2.
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.
R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065
Review URL: https://webrtc-codereview.appspot.com/50679004
Cr-Commit-Position: refs/heads/master@{#8985}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ad1f9b61a3107ca27ee023990dc576abc38f05ac |
|
08-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove warning on input frames before config.
Removes log spam for AppRTC when only one client is connected.
BUG=4512
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48019005
Cr-Commit-Position: refs/heads/master@{#8947}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 64c1e8cda5cb4db85c5c296bf2f6a8181af7de9d |
|
02-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8917}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 31331cfd2d3d17958942b67190c8b943c05b084f |
|
01-Apr-2015 |
Minyue <minyue@webrtc.org> |
Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae.
Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.
Review URL: https://webrtc-codereview.appspot.com/47909004
Cr-Commit-Position: refs/heads/master@{#8911}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1b1c15cad16de57053bb6aa8a916079e0534bdae |
|
01-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.
Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome
BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47399004
Cr-Commit-Position: refs/heads/master@{#8905}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 23914fe756903353eae13fffc868d2c84f51f06f |
|
31-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject RTP one-byte extension ID 0.
Only accept local identifiers in the range 1-14 inclusive.
BUG=1788, chromium:471328
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50549004
Cr-Commit-Position: refs/heads/master@{#8900}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| dfd53fe26b013d0948024a38eec6fbc31c29a094 |
|
27-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Raise streams for SetMaxSendBitrates above 2000k.
Fixes b=AS effectively not setting bitrates above 2000k.
BUG=1788,4469
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47839004
Cr-Commit-Position: refs/heads/master@{#8882}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 74d9ed7d853677d297807021436467a4f97584ac |
|
26-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Report send codec name in GetStats().
BUG=4461
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51439004
Cr-Commit-Position: refs/heads/master@{#8869}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d6f4c25eedcfd502920f1b2a24744badd9da80be |
|
26-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject streams reusing simulcast or RTX SSRCs.
BUG=1788, chromium:470122, chromium:470856
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42919004
Cr-Commit-Position: refs/heads/master@{#8868}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| d4362cd3368d5fe542911c375b3a5c9f24b2f29d |
|
25-Mar-2015 |
Peter Boström <pbos@webrtc.org> |
Reject StreamParams with RTX SSRCs not in ssrcs.
BUG=1788, chromium:470122
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44859004
Cr-Commit-Position: refs/heads/master@{#8855}
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| eebcab5ce99d3e8641dd92a569916b0d24e29fca |
|
24-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
rtc::Buffer: Rename length to size, for conformance with the STL
And add a constructor for creating an uninitialized Buffer of a
specified size.
(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48579004
Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| a5f6fb53ba802fcf44e3c187d798c5a53ad555df |
|
23-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Permit single-stream max bitrates above 2000k.
BUG=4463
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49509004
Cr-Commit-Position: refs/heads/master@{#8839}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8839 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b9557a9bb7ed5f9aa1e7b3a64de4238572794ae3 |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix code to handle crashes for non-VP8.
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.
BUG=1667, 1788
R=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44789004
Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8296ec518b2659de922668bfe0db57e71eb17e74 |
|
20-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44779004
Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| af612d5e0769571544952cbe55e675748afa9bdd |
|
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
Original cl description:
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/.
Patchset 1 contains the original patch after rebase.
Patshet 2 fix webrtc_perf_tests reported in chromium:465306
Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47629004
Cr-Commit-Position: refs/heads/master@{#8776}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| afdd5dd372d69be7244a3d90d70de9d5ecd60eb9 |
|
12-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame""
This reverts r8683 and is a reland of r8682.
Reason for revert: The thread checker in Chromium that crashed has been fixed now.
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/40319004
Cr-Commit-Position: refs/heads/master@{#8696}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b218ff553148b9a26c82e3b3a46d626c4438cedd |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"
This reverts r8682.
Reason for revert: Fails on Chromium FYI content_browsertests
BUG=1128
TBR=tommi,pbos,pthatcher
Review URL: https://webrtc-codereview.appspot.com/47529004
Cr-Commit-Position: refs/heads/master@{#8683}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 370a72cc3ff928099c6ec6766659ed12155b74df |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy from cricket::VideoFrame to I420VideoFrame
BUG=1128
R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42249004
Cr-Commit-Position: refs/heads/master@{#8682}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d7452a016812ab1de69c3d7a53caca5b06c64990 |
|
10-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."
This reverts commit r8633.
Reason for revert: Performance regressions in browser_tests_new_vie and webrtc_perf_tests.
BUG=1128,chromium:465287,chromium:465306
TBR=pbos,mflodman,perkj
Review URL: https://webrtc-codereview.appspot.com/46549004
Cr-Commit-Position: refs/heads/master@{#8670}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8670 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| a2a6fe66a39797ea61a04d80ce3afc494d850bfc |
|
06-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Reconfigure default streams on AddRecvStream.
Makes sure RTX can be used for streams that have received early media
before being properly configured.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46499004
Cr-Commit-Position: refs/heads/master@{#8634}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| bcead305a2f27c30c72c6a3824fdf12f4b83c2eb |
|
06-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.
This removes the none const pointer entry and SwapFrame.
Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker.
Also, the video engine must ensure that time stamps are always increasing.
With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame
BUG=1128
R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46429004
Cr-Commit-Position: refs/heads/master@{#8633}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8633 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 2386d6dd92f10a715f131b5ad408b1babc1f35b0 |
|
05-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert 8599 "Revert 8580 "Unify underlying frame buffer in I420VideoFrame and...""
It's possible to build Chrome on Windows with this patch now.
BUG=1128
> This is unfortunately causing build problems in Chrome on Windows.
>> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>>
>> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>>
>> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>>
>> Some additional minor changes are:
>> * Disallow creation of 0x0 texture frames.
>> * Remove the half-implemented ref count functions in I420VideoFrame.
>> * Remove the Alias functionality in WebRtcVideoFrame
>>
>> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
>> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
>> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>>
>> BUG=1128
>> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>>
>> Review URL: https://webrtc-codereview.appspot.com/42469004
R=pbos@webrtc.org
TBR=mflodman, pbos, perkj, tommi
Review URL: https://webrtc-codereview.appspot.com/45489004
Cr-Commit-Position: refs/heads/master@{#8616}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8616 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1f94407319f85abc286c993774a4ea93807ec32e |
|
04-Mar-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8580 "Unify underlying frame buffer in I420VideoFrame and..."
This is unfortunately causing build problems in Chrome on Windows.
> Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
>
> Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
>
> This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
>
> Some additional minor changes are:
> * Disallow creation of 0x0 texture frames.
> * Remove the half-implemented ref count functions in I420VideoFrame.
> * Remove the Alias functionality in WebRtcVideoFrame
>
> The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
> * Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
> * Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
>
> BUG=1128
> R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/42469004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42199005
Cr-Commit-Position: refs/heads/master@{#8599}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8599 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 058b1f17ac43b1fe69a8c18aaa7999ba88733dfd |
|
04-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove GetReceiveBandwidthEstimatorStats.
Removes unnecessary non-standard stats that we don't really make use of.
BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47379004
Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c8895aa2f31e05d3bd4d29507af3bbfcaa638499 |
|
03-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Unify underlying frame buffer in I420VideoFrame and WebRtcVideoFrame
Currently, I420VideoFrame uses three webrtc::Plane to store pixel data, and WebRtcVideoFrame uses WebRtcVideoFrame::FrameBuffer/webrtc::VideoFrame. The two subclasses WebRtcTextureVideoFrame and TextureVideoFrame use a NativeHandle to store pixel data, and there is also a class WebRtcVideoRenderFrame that wraps an I420VideoFrame.
This CL replaces these classes with a new interface VideoFrameBuffer that provides the common functionality. This makes it possible to remove deep frame copies between cricket::VideoFrame and I420VideoFrame.
Some additional minor changes are:
* Disallow creation of 0x0 texture frames.
* Remove the half-implemented ref count functions in I420VideoFrame.
* Remove the Alias functionality in WebRtcVideoFrame
The final goal is to eliminate all frame copies, but to limit the scope of this CL, some planned changes are postponed to follow-up CL:s (see planned changes in https://webrtc-codereview.appspot.com/38879004, or https://docs.google.com/document/d/1bxoJZNmlo-Z9GnQwIaWpEG6hDlL_W-bzka8Zb_K2NbA/preview). Specifically, this CL:
* Keeps empty subclasses WebRtcTextureVideoFrame and TextureVideoFrame, and just delegates the construction to the superclass.
* Keeps the deep copies from cricket::VideoFrame to I420VideoFrame.
BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42469004
Cr-Commit-Position: refs/heads/master@{#8580}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8580 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f1f0d9a4cd53f4eacbf791cb7317612fa7382a45 |
|
02-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEngine::SetVoiceEngine.
Instead enforcing that a voice engine is set on construction. Apart from
simplifying the class this permits tracing to be set up in the
constructor without worrying about racing sets from SetVoiceEngine
later.
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44489004
Cr-Commit-Position: refs/heads/master@{#8555}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8555 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 2a72c6506a49b15d5e079eaa28cb80abb445684b |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Keep feedback params in SetDefaultEncoderConfig.
Prevents NACK etc. from breaking completely as it won't be reported in
the generated SDP.
BUG=1788
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40109004
Cr-Commit-Position: refs/heads/master@{#8519}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8519 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 891d48393e5ccd2f5e03d509c544c00a3d88cbbc |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up target_media_bitrate in VideoSendStream.
Also wires up target_enc_bitrate in WebRtcVideoEngine2.
BUG=1667,1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42479004
Cr-Commit-Position: refs/heads/master@{#8515}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8515 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3e6e271ec3253e78ae0eb72156e5236d43f8731d |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement CpuOveruseMetrics as callbacks.
Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and
corresponding stats to VideoSendStream::Stats.
BUG=1667, 1788
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42429004
Cr-Commit-Position: refs/heads/master@{#8513}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9a4410e9934578e84cc129b978a29e151d957994 |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement adaptation stats in WebRtcVideoEngine2.
BUG=1788
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42489004
Cr-Commit-Position: refs/heads/master@{#8510}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8510 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 09c77b95bb62566be64da662f0b3b6a838ec6553 |
|
25-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add decoder-timing stats to VideoReceiveStream.
Also breaks out SsrcStats from VideoReceiveStream::Stats as they don't
have that much overlap.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667, 1788
Review URL: https://webrtc-codereview.appspot.com/40819004
Cr-Commit-Position: refs/heads/master@{#8501}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8501 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d324546ced76d4e792338af4f7d02a5cd8819f92 |
|
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ :
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 77e11bbe834e3b096db57278d2ad7c76d8c26d66 |
|
23-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up preferred/nominal_bitrate to stats.
Also adds a test that shows that actual_enc_bitrate was not summed
correctly plus fixing it.
Additionally reducing locking when grabbing stats.
BUG=1778
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34319004
Cr-Commit-Position: refs/heads/master@{#8464}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8464 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1ed6224eafc7816f25d1906e4d709afdf2ad8f0f |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert r8430 "Remove dead stats from Video{Sender,Receiver}Info."
This breaks compilation outside this codebase that needs to have it
removed before.
BUG=4322
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42009004
Cr-Commit-Position: refs/heads/master@{#8432}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8432 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8ad05b76281e73f92051125aee81d85227c6a9bc |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove dead stats from Video{Sender,Receiver}Info.
These stats are neither filled nor plumbed further and might as well be
removed (as proven by how easy they were to remove).
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39219004
Cr-Commit-Position: refs/heads/master@{#8430}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8430 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/37489004
Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b4987bfc24e1e755a6c54053d09a58d1e72228bb |
|
18-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Send black frame with previous size when muting.
Instead of sending a black frame that's the size of the VideoFormat send
a black frame in the format we're already sending. This prevents
expensive encoder reconfiguration when the sending format is a different
resolution. This speeds up setting a null capturer (removing the
capturer) significantly as it doesn't entail an encoder reconfiguration.
R=mflodman@webrtc.org, pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/39179004
Cr-Commit-Position: refs/heads/master@{#8405}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8405 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e9facf8bb32a1688f2156009c755caa2904e1ac9 |
|
17-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Add range checks in a variety of places where the values will subsequently be
expected to be 0-127.
BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika
Review URL: https://webrtc-codereview.appspot.com/37759004
Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 86196c4f481d7f515e54806988f763169e8b9206 |
|
16-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Setup encoders inexpensively before first frame.
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ff689be3c0c59c1be29aaa0697aa0f762566d6c6 |
|
12-Feb-2015 |
andresp@webrtc.org <andresp@webrtc.org> |
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35079004
Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0d852d5c27a759fe7aadc500bd7b3cadfae3deb8 |
|
09-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Use VideoReceiveStream as an ExternalRenderer.
Removes AddRenderCallback from ViERenderer and implements
VideoReceiveStream on top of DeliverI420Frame like WebRtcVideoEngine
currently does today.
Also adds ::IsTextureSupported() to the VideoRenderer interface to
permit querying whether an external renderer supports texture rendering.
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/34169004
Cr-Commit-Position: refs/heads/master@{#8299}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8299 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 5e161616b17900c06809e7275afca96363d44ad5 |
|
30-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove CPU monitor from WebRtcVideoEngine2.
CPU adaptation is based on timings done inside webrtc, not actual CPU
values anymore. This code has never been wired up and is causing flakes
on at least valgrind, but possibly also on actual platforms.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34089004
Cr-Commit-Position: refs/heads/master@{#8221}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8221 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 50fe359eb614e1bbe41124b9c19263019da0395d |
|
29-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add tracing for slow paths in new video API.
Allows tracking what actually takes time in SetRemoteDescription and
SetLocalDescription.
BUG=1788
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38809004
Cr-Commit-Position: refs/heads/master@{#8202}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8202 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| fc5ad95fecc5ddc7d98dcfbac1c4e75a7814253f |
|
27-Jan-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139
Link to original CL: https://review.webrtc.org/36909004/
R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227
Review URL: https://webrtc-codereview.appspot.com/39669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0f988447496e5d656d52bea279c8511d3569cb11 |
|
23-Jan-2015 |
tkchin@webrtc.org <tkchin@webrtc.org> |
Revert 8139 "Implement elapsed time and capture start NTP time e..."
> Implement elapsed time and capture start NTP time estimation.
>
> These two elements are required for end-to-end delay estimation.
>
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36909004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ad3ee2c46bf502a18847229d42dd081c9e753c70 |
|
23-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement elapsed time and capture start NTP time estimation.
These two elements are required for end-to-end delay estimation.
BUG=1788
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8139 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| cfd82dfc1156f6610388bec0ebbdeacaf47e9719 |
|
22-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
Prepares for adding FEC bytes to the StreamDataCounter.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f1c8b905204bc7a6c74271ead038f5d80d8d3eed |
|
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove WebRtcVideoEncoderFactory2.
This interface is no longer required and just adds complexity.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/33009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f18fba2f7b3d1fad7b7b38a9a5dc281bef06c50e |
|
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement SimulcastEncoderAdapter support.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/37589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8f27fcce79584378da97f0d84574564799e138d6 |
|
09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..."
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.
> Support associated payload type when registering Rtx payload type.
>
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
>
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26259004
>
> Patch from Changbin Shao <changbin.shao@intel.com>.
TBR=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 2a169640a3225a559f926fe74f1fe1af239e191f |
|
09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type.
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.
BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26259004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c37e72e890cb1c769af9006dbd2e582c1a2e2a50 |
|
05-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Make setting identical RTP extensions a no-op.
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.
BUG=1788,4077
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 896888b7e4cea97c65786b0e63bf2f65dc7d2390 |
|
02-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove min bitrate from simulcast streams.
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/38529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 46d4d29a751c559b6f01b311a1e4aa14a2586a46 |
|
23-Dec-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Add field trial for screenshare bitrates when using temporal layers.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 742386a13670337db6e3bbf4cf54e7cb24a9b717 |
|
19-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Enable payload-based padding by default and remove the API.
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e575e9c40f7e2aeb28486f6e4b96910bc744c7ec |
|
14-Dec-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| a85307737cc9ea3e79b86daf96d455fca4ad1bb4 |
|
10-Dec-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 81702493-> 81755413
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 599e299b9dc3dc07fc78cfeaba629566a201b4f1 |
|
05-Dec-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
cricket::VideoFrame int64 to int64_t.
Needed for successful compile of ios arm64.
BUG=3898
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30359004
Patch from Zeke Chin <tkchin@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0fb6ad2004d3b86cb912c93a773e3f9162392e54 |
|
03-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Check if cpu_monitor_ exists before Stop().
R=asapersson@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/25279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 273a414b0ec2e58fdf3b817ad8b1a02f4ce15287 |
|
01-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report encoded frame size in VideoSendStream.
Implements reporting transmitted frame size in WebRtcVideoEngine2.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033
Review URL: https://webrtc-codereview.appspot.com/33399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| bdcf38c89446b1b464a646414f6cd7573a190bd1 |
|
21-Nov-2014 |
magjed@webrtc.org <magjed@webrtc.org> |
cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class
There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc
R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d819803d4570564a9800a7dd54f4593e6e21a6e7 |
|
10-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up DSCP support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/24249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 957e802fe0e6e765425955cc1e3e02f73d1a670b |
|
10-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor SetDefaultEncoderConfig to work on existing codecs.
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/32009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 188d3b2245b49f21468840386d81b080176b434b |
|
07-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Enable VP9 video codec support on webrtcvideoengine behind a field trial.
BUG=chromium:431285
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| a2ef4fe9c331e7668b9e8ff64ce5141a535a5f21 |
|
07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Prevent a lot of VideoSendStream reconfigures.
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 82775b13965b4d41299b097c09c30c4ab160cdac |
|
07-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime.
This will allow to plugin VP9 based on a field trial.
R=pbos@webrtc.org, pbos, pthatcher
Review URL: https://webrtc-codereview.appspot.com/27949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 19b47410044aecac20f3f46a4d207018fc466e2e |
|
06-Nov-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Removing unused method GetDefaultVideoEncoderConfig.
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 88ef6322864b4071df4ed724a3989a9183d92172 |
|
04-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Falling back on single-stream on multiple SSRC.
Instead of failing, use one stream. Also clamp video min bitrate.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 96a93259b361f4b03080a188d781b0835cf4edaf |
|
03-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement external decoder support in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b7ed7799e77d3b315f5016951ecb90d18f10fdcb |
|
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement conference-mode temporal-layer screencast.
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667
Review URL: https://webrtc-codereview.appspot.com/23269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3bf3d238c8c4578e444e5a601684db68c79a29ca |
|
31-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Configure A/V sync in WebRtcVideoEngine2.
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/23249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 776e6f289c7396a1143b8b36b03f88b08ac8cba3 |
|
29-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use external VideoDecoders in VideoReceiveStream.
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.
Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.
Additionally addresses a data race in VideoReceiver that was exposed with this change.
R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667
Review URL: https://webrtc-codereview.appspot.com/27829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ae694effd85d501f15600275dec96522a00c4feb |
|
28-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 78642371-> 78680406
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| efc82c2c734171faba9e400ff60a114e7af2ebcc |
|
27-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement screencast settings for WebRtcVideoEngine2.
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| fa553ef6053b20f3768d5fe4314e8c993648bf0a |
|
20-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set up start bitrate in WebRtcVideoEngine2.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/27789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1ecbe45c7e4c9142896cb2810d699558518f4f28 |
|
14-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77689511-> 77696841
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 7fe1e03dd6da66401010119734245f114bf06645 |
|
14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up external encoders.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/30649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3c16d8bd1c0a3eea94a6678497eae0cf8e7f0187 |
|
13-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77414393-> 77554188
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 97abeee2825ac93b62397feea74d0ad02d42540d |
|
09-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77263371-> 77296420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 575d126a3d4a4bf6d43ea07189ac201f6bfe0798 |
|
08-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Protect send_/recv_streams_ in WebRtcVideoEngine2.
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.
R=stefan@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/22959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 963b979510f6521fd69576f146235c6a5c0f8264 |
|
07-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove potential deadlock in WebRtcVideoEngine2.
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.
R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999
Review URL: https://webrtc-codereview.appspot.com/26729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| d60d79a14594cbc8266e4a50391ddbe64ed491f0 |
|
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Thread annotation of rtc::CriticalSection.
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.
This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.
R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0a2087a7110e2455ce68f2c85068df5ae447508f |
|
23-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Skeleton for registering external encoders/decoders.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/31429005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 83f95ba9a645099df5e19a91030029181d766b40 |
|
22-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove engine-level SetOptions.
Already removed in WebRtcVideoEngine.
R=andresp@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 933d88af58b00517570ef78f38852bfd7fb1bb02 |
|
18-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 75818332-> 75837294
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 6cd6ba8ae016200a7a13b43294b8faf5d1d4affd |
|
18-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Expose VP8/H264 defaults through video_encoder.h.
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=3070
Review URL: https://webrtc-codereview.appspot.com/19299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ab990ae43a2b84b103cb3c50bc38502375c13e68 |
|
17-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.
BUG=3070
TBR=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| cddd17c0f89cfaa9d2f21118ae90b45dae3b4aee |
|
16-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Recreate VideoStreams when setting resolution.
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/29469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 307d3dbdeed71d42edf38d3828081b11a5a416fb |
|
11-Sep-2014 |
henrikg@webrtc.org <henrikg@webrtc.org> |
Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."
Speculative revert, seems to be reason for flaky Win FYI bot compile break.
> Expose VideoEncoders with webrtc/video_encoder.h.
>
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
>
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/21929004
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b420191743fc135222c862deeaa4cf9dec249fe3 |
|
09-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Expose VideoEncoders with webrtc/video_encoder.h.
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.
BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 992febb9978d2ded1a2c3c8a42ea18ee071ca3ae |
|
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74873066-> 74873164
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 818b7b3ac982e9e1f579904c5f160103da046dcf |
|
05-Sep-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 74825084-> 74825992
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| c4175b9fdf7d23eb58a044ff39e2b096f9091995 |
|
03-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Set resolution based on incoming VideoFrames.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/17269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b648b9d85c5d07b0866ef45f5be587f71b0849b4 |
|
26-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove test constructor in WebRtcVideoEngine2.
Removes the need for ::Construct().
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3740d741068698baf987b1ced5ea485378e16d04 |
|
23-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73927658-> 73927775
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 97fdeb8329cf5c328fa531c0a61c3dd181eb4833 |
|
22-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove static initializer in WebRtcVideoEngine2.
Blocks import into chromium.
R=tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/18249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6954 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| ef8bb8d9b0bca0b1fd1ddb0a17df665e9dfaf9ad |
|
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure that muting muted streams succeeds.
We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 432893a1002636aa83f7d356ed8e6f80f908d134 |
|
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove TODO saying to remove WebRtcVideoFrame.
Comment was added prematurely, there's no decision to get rid of this
type.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| a09a99950ec40aef6421e4ba35eee7196b7a6e68 |
|
13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| afb554f404d68e6f3ca5395216f776169370713d |
|
13-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move default-recv-channels to a separate class.
BUG=1788,3099
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c3d2bd28a3e8badc434a5081dd36f4ac41b4e3f2 |
|
12-Aug-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix GetStats() crash.
GetStats() can be called before codecs are set and the underlying
webrtc::VideoSendStream is created, leading to a null-pointer
dereference.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6876 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d4e598d57aed714a599444a7eab5e8fdde52a950 |
|
29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 257e130a1639febeb3ffc4d42943be3cb58151c7 |
|
25-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Set NACK/REMB when setting receive codecs.
Enabling an additional test to ensure that REMB can be both enabled and
disabled by setting VideoCodecs.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6785 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 9359cb3e75c7100dab4c687f60dd28dc613280e4 |
|
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable SendAndReceive tests.
Also fixes a crash in ::SetCapturer which wasn't exposed by tests
before.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6765 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 5ff71ab4b369fe3dbfaec5f91cd2e491397eff33 |
|
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "(Auto)update libjingle 71675033-> 71726409"
This reverts commit r6761 which looks like an accidental auto-revert of
r6760.
BUG=1788
TBR=wu@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6763 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f67f6aa74187dbb804ec3bc98b9551db9fcf5571 |
|
23-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 71675033-> 71726409
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6761 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8120353342f27df70018a808efa92acc8a07d9f2 |
|
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement suspend-below-min-bitrate option.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6760 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 543e589205af006f6b999a2c5df51d3fb722d925 |
|
23-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up VideoOptions for payload-based padding.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6759 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 6f48f1bf68a10669c9bcd81262c1a98ed2a8d462 |
|
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement encoder options in WebRtcVideoEngine2.
Implementing default options to enable denoising by default and wiring
up encoder settings to propagate VP8 settings.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| cadd078cf994b873f344621901fe62a621bbaa6c |
|
22-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused config.h and math.h includes.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6756 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 3c10758b3bb9519d5e582c00f454ac30196ac4e7 |
|
20-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check before send/receive rtp header extensions.
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13949004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 8fdeee6abfcb560233b5e769afb1c1c72cc2100d |
|
20-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement Base::ConstrainNewCodec2.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6743 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e6f84ae8a602ce78733d20b280ce32198e7ecef5 |
|
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial WebRtcVideoEngine2::GetStats().
Also forward-declaring and moving WebRtcVideoRenderer out of header.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d1ea06b3d5adab352741df5092c56b20f3e1a74f |
|
18-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Restart VideoReceiveStreams in WebRtcVideoEngine2.
Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.
Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.
BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 5301b0f1fce9478dfa56476e174332a1d67b053a |
|
17-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move additional state into WebRtcVideoSendStream.
Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| cb859ecd3b9435633434ca3c028eb60c8e8c5938 |
|
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace strcpy with talk_base::strcpyn.
Cpplint reports error 'Almost always, snprintf is better than strcpy'
when checking code styles. The function talk_base::strcpyn() is a better
option than strcpy().
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12919004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6686 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| df9bbbee56f4d9ecef93b3c46964b6f29803f81b |
|
19-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69567902-> 69568113
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6498 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 587ef60056ff0e301a95a9eb8231fb0cae6b69b1 |
|
16-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement RTP extension support in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| f99c2f2dbcaab24b45295cb9e06c3c52ad349d81 |
|
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add NACK feedback parameter to WebRtcVideoEngine2.
Also fixing enabling/disabling of NACK. Previous implementation was made
under the assumption that NACK should always be enabled which caused
both missing NACK settings in SDP as well as broken interop between old
and new WebRtcVideoEngines.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6431 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| e322a175f6f38c4ed39296d9724edf005e536a63 |
|
13-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement RTX tests+fixes in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6430 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d41eaeb7cded2b2cda83f53aa320cf18e2d07380 |
|
12-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69005149-> 69049090
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6408 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 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/talk/media/webrtc/webrtcvideoengine2.cc
|
| 0d523eea831e616c415c61765127ed5eb17e5f11 |
|
05-Jun-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove static initializer from WebRtcVideoEngine2.
BUG=
R=pliard@google.com, pthatcher@webrtc.org, pliard@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| c34bb3a88627672d99b1c037d36dbeb23407fae4 |
|
30-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Log default receive stream creation.
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 198647473ba207d59dc94216ef38496d43d15592 |
|
30-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Implement and fix new-API NackIsEnabled test.
Required enabling NACK on receiver side which was apparently missed.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16499007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 1e019d10b8bcd96e8cf6b3d3df2730449fbed939 |
|
16-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix delivery error-checking missed in r6151.
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.
BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 4e545cc24478df6dec0f73cb8f5b9e5720fbce59 |
|
14-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update webrtcvideoengine2.cc to use DeliveryStatus.
talk/ changes corresponding to https://review.webrtc.org/12289005/.
BUG=3228
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| b5a22b14648c53874b4b76368a1a2271d985e875 |
|
13-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r6110 and r6109.
Effectively re-landing r6104 as well as r6108 which includes a fix to a
compile error that caused r6104 to be reverted in r6110.
BUG=
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6119 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| 17911dca8099707b5c050741a108b95b79a4da66 |
|
12-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66798415-> 66813165
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6110 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|
| d266a2020f9e86a787eada77d458ee75426d68af |
|
12-May-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Initial wiring of new webrtc API in libjingle.
BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org
TBR=juberti@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6104 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/webrtcvideoengine2.cc
|