436 Commits

Author SHA1 Message Date
guidou
716d7ac5c1 Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ )
Reason for revert:
Suspect of breaking Chrome FYI bots.

See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder

Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
 #include "third_party/webrtc/video_encoder.h"
                                              ^

Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570

TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881

Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
2017-04-03 16:15:52 +00:00
ilnik
c42f540570 Move video_encoder.h and video_decoder.h to /api and create GN targets for them
BUG=webrtc:5881
# Because PRESUBMIT ignores LINT blacklist for moved files and these
# headers have some not easy to resolve issues.
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2780943003
Cr-Commit-Position: refs/heads/master@{#17511}
2017-04-03 15:37:32 +00:00
sprang
c5d62e29ca Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2783183003/ )
Reason for revert:
Seem to be a flaky test rather than an issue with this cl. Creating reland, will add code to reduce flakiness to that test.

Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
>
> Reason for revert:
> This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780
>
> Original issue's description:
> > Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
> >
> > Reason for revert:
> > Found issue with test case, will add fix to reland cl.
> >
> > Original issue's description:
> > > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> > >
> > > Reason for revert:
> > > Breaks perf tests:
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> > >
> > > Original issue's description:
> > > > Add framerate to VideoSinkWants and ability to signal on overuse
> > > >
> > > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > > current degradation preference is maintain-resolution rather than
> > > > balanced.
> > > >
> > > > BUG=webrtc:4172
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2716643002
> > > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > > Committed: 72acf25261
> > >
> > > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2764133002
> > > Cr-Commit-Position: refs/heads/master@{#17331}
> > > Committed: 8b45b11144
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2781433002
> > Cr-Commit-Position: refs/heads/master@{#17474}
> > Committed: 3ea3c77e93
>
> TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2783183003
> Cr-Commit-Position: refs/heads/master@{#17477}
> Committed: f9ed235c9b

R=ilnik@webrtc.org,stefan@webrtc.org
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2789823002
Cr-Commit-Position: refs/heads/master@{#17498}
2017-04-03 06:53:04 +00:00
lliuu
f9ed235c9b Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #8 id:410001 of https://codereview.webrtc.org/2781433002/ )
Reason for revert:
This has resulted in failure of CallPerfTest.ReceivesCpuOveruseAndUnderuse test on the Win7 build bot https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1780

Original issue's description:
> Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
>
> Reason for revert:
> Found issue with test case, will add fix to reland cl.
>
> Original issue's description:
> > Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
> >
> > Reason for revert:
> > Breaks perf tests:
> > https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> > https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
> >
> > Original issue's description:
> > > Add framerate to VideoSinkWants and ability to signal on overuse
> > >
> > > In ViEEncoder, try to reduce framerate instead of resolution if the
> > > current degradation preference is maintain-resolution rather than
> > > balanced.
> > >
> > > BUG=webrtc:4172
> > >
> > > Review-Url: https://codereview.webrtc.org/2716643002
> > > Cr-Commit-Position: refs/heads/master@{#17327}
> > > Committed: 72acf25261
> >
> > TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2764133002
> > Cr-Commit-Position: refs/heads/master@{#17331}
> > Committed: 8b45b11144
>
> TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2781433002
> Cr-Commit-Position: refs/heads/master@{#17474}
> Committed: 3ea3c77e93

TBR=ilnik@webrtc.org,stefan@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2783183003
Cr-Commit-Position: refs/heads/master@{#17477}
2017-03-30 17:44:38 +00:00
sprang
3ea3c77e93 Reland of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #1 id:1 of https://codereview.webrtc.org/2764133002/ )
Reason for revert:
Found issue with test case, will add fix to reland cl.

Original issue's description:
> Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
>
> Reason for revert:
> Breaks perf tests:
> https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
> https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325
>
> Original issue's description:
> > Add framerate to VideoSinkWants and ability to signal on overuse
> >
> > In ViEEncoder, try to reduce framerate instead of resolution if the
> > current degradation preference is maintain-resolution rather than
> > balanced.
> >
> > BUG=webrtc:4172
> >
> > Review-Url: https://codereview.webrtc.org/2716643002
> > Cr-Commit-Position: refs/heads/master@{#17327}
> > Committed: 72acf25261
>
> TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2764133002
> Cr-Commit-Position: refs/heads/master@{#17331}
> Committed: 8b45b11144

TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,skvlad@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2781433002
Cr-Commit-Position: refs/heads/master@{#17474}
2017-03-30 14:23:48 +00:00
nisse
e5ad5ca06a Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #1 id:1 of https://codereview.webrtc.org/2784543002/ )
Reason for revert:
Intend to fix perf failures and reland.

Original issue's description:
> Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
>
> Reason for revert:
> Reverting since this seems to break multiple WebRTC Perf buildbots
>
> Original issue's description:
> > Don't hardcode MediaType::ANY in FakeNetworkPipe.
> >
> > Instead let each test set the appropriate media type. This simplifies
> > demuxing in Call and later in RtpTransportController.
> >
> > BUG=webrtc:7135
> >
> > Review-Url: https://codereview.webrtc.org/2774463003
> > Cr-Commit-Position: refs/heads/master@{#17418}
> > Committed: 9c47b00e24
>
> TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2784543002
> Cr-Commit-Position: refs/heads/master@{#17427}
> Committed: 3a3bd50610

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,lliuu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2783853002
Cr-Commit-Position: refs/heads/master@{#17459}
2017-03-30 06:57:43 +00:00
sprang
fe627f30cb Use simulcast for screenshare only in conference mode
Also, don't crash if InitEncode fails for vp8.

BUG=chromium:705505

Review-Url: https://codereview.webrtc.org/2779163002
Cr-Commit-Position: refs/heads/master@{#17452}
2017-03-29 15:24:59 +00:00
dminor
588101cb91 Change minimum DTMF event duration to be 40 milliseconds
The current value of 100 milliseconds is the recommended default value
in https://w3c.github.io/webrtc-pc/#rtcdtmfsender; the actual minimum specified is 40 milliseconds.

BUG=webrtc:7163

Review-Url: https://codereview.webrtc.org/2699503002
Cr-Commit-Position: refs/heads/master@{#17430}
2017-03-28 18:18:32 +00:00
lliuu
3a3bd50610 Revert of Don't hardcode MediaType::ANY in FakeNetworkPipe. (patchset #4 id:60001 of https://codereview.webrtc.org/2774463003/ )
Reason for revert:
Reverting since this seems to break multiple WebRTC Perf buildbots

Original issue's description:
> Don't hardcode MediaType::ANY in FakeNetworkPipe.
>
> Instead let each test set the appropriate media type. This simplifies
> demuxing in Call and later in RtpTransportController.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/2774463003
> Cr-Commit-Position: refs/heads/master@{#17418}
> Committed: 9c47b00e24

TBR=stefan@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,sprang@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2784543002
Cr-Commit-Position: refs/heads/master@{#17427}
2017-03-28 16:40:59 +00:00
solenberg
83862e3c14 Remove VoECodec from FakeWebRtcVoiceEngine.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2785443002
Cr-Commit-Position: refs/heads/master@{#17420}
2017-03-28 12:07:15 +00:00
nisse
9c47b00e24 Don't hardcode MediaType::ANY in FakeNetworkPipe.
Instead let each test set the appropriate media type. This simplifies
demuxing in Call and later in RtpTransportController.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774463003
Cr-Commit-Position: refs/heads/master@{#17418}
2017-03-28 11:59:41 +00:00
minyue
cecec1060f Set max bitrate for audio send stream based on RtpParameters.
BUG=webrtc:7392

Review-Url: https://codereview.webrtc.org/2775483004
Cr-Commit-Position: refs/heads/master@{#17399}
2017-03-27 20:04:25 +00:00
jianj
a5e8aa6e2b Vp9: Enable denoiser by default.
BUG=webrtc:7412

Review-Url: https://codereview.webrtc.org/2765663002
Cr-Commit-Position: refs/heads/master@{#17395}
2017-03-27 17:09:00 +00:00
kwiberg
1c07c70d88 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2774833003
Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 14:15:49 +00:00
stefan
1ccf73f830 Fix issue with conflicting behavior if setting a max BW with b=AS on both audio and video.
This reverts to previous behavior where b=AS only affects the codec bitrate for audio streams, and not the max bandwidth estimate.

BUG=chromium:703903

Review-Url: https://codereview.webrtc.org/2774123002
Cr-Commit-Position: refs/heads/master@{#17386}
2017-03-27 10:51:18 +00:00
kwiberg
670a7f3611 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
Reason for revert:
Makes perf and Chromium FYI bots unhappy.

Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba

TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
2017-03-24 12:56:21 +00:00
kwiberg
1724cfbdba WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
This removes one more place where we were unable to handle codecs not
in the built-in set.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
2017-03-24 10:16:04 +00:00
skvlad
8b45b11144 Revert of Add framerate to VideoSinkWants and ability to signal on overuse (patchset #14 id:250001 of https://codereview.webrtc.org/2716643002/ )
Reason for revert:
Breaks perf tests:
https://build.chromium.org/p/client.webrtc.perf/builders/Win7/builds/1679
https://build.chromium.org/p/client.webrtc.perf/builders/Android32%20Tests%20%28L%20Nexus5%29/builds/2325

Original issue's description:
> Add framerate to VideoSinkWants and ability to signal on overuse
>
> In ViEEncoder, try to reduce framerate instead of resolution if the
> current degradation preference is maintain-resolution rather than
> balanced.
>
> BUG=webrtc:4172
>
> Review-Url: https://codereview.webrtc.org/2716643002
> Cr-Commit-Position: refs/heads/master@{#17327}
> Committed: 72acf25261

TBR=nisse@webrtc.org,magjed@webrtc.org,kthelgason@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2764133002
Cr-Commit-Position: refs/heads/master@{#17331}
2017-03-21 20:26:06 +00:00
sprang
72acf25261 Add framerate to VideoSinkWants and ability to signal on overuse
In ViEEncoder, try to reduce framerate instead of resolution if the
current degradation preference is maintain-resolution rather than
balanced.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2716643002
Cr-Commit-Position: refs/heads/master@{#17327}
2017-03-21 18:54:11 +00:00
solenberg
22818a5804 (Re)move VoE specific enums from common_types.h.
BUG=webrtc:5876

Review-Url: https://codereview.webrtc.org/2745253007
Cr-Commit-Position: refs/heads/master@{#17267}
2017-03-16 08:20:23 +00:00
solenberg
9a5f032227 Remove VoEHardware interface usage.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2737633002
Cr-Commit-Position: refs/heads/master@{#17248}
2017-03-15 13:14:12 +00:00
solenberg
ebb349d7c9 Revert to allowing only 1 unsignaled receive stream for audio.
Reason to go back is that we may end up with a bunch of streams that are never cleaned up and consume resources.

BUG=webrtc:7175, b/35863246

Review-Url: https://codereview.webrtc.org/2746763002
Cr-Commit-Position: refs/heads/master@{#17210}
2017-03-13 12:46:15 +00:00
pbos
5c7760a3a2 Support removing b=AS bandwidth constraints.
In code this is represented as setting -1 to max_bandwidth_bps, but this
value was being ignored by webrtcvideoengine2.cc, so previous restrictions
would still apply.

BUG=webrtc:6202
TEST=Setting "unlimited" for Bandwidth in Chromium in https://webrtc.github.io/samples/src/content/peerconnection/bandwidth/.
R=deadbeef@webrtc.org,stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2740783006
Cr-Commit-Position: refs/heads/master@{#17175}
2017-03-10 19:23:12 +00:00
nisse
c69385de8b Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
Use of FlexFEC is known when streams are created in
WebRtcVideoChannel2, so this replaces the code in Call to infer
FlexFEC config of video streams from the configuration of the FlexFEC
stream(s). This also allows us to switch to a more logical creation
order, where media streams are created before the FlexFEC stream.

This is done in preparation for a larger refactoring of the RTP
demuxing done in Call.

BUG=None

Review-Url: https://codereview.webrtc.org/2712683002
Cr-Commit-Position: refs/heads/master@{#17143}
2017-03-09 14:13:20 +00:00
kjellander
a8d8aadba8 Refactor + enable GN check on video_coding_utility
To avoid the cyclic dependency

BUG=webrtc:6828
NOTRY=True
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2717113002
Cr-Commit-Position: refs/heads/master@{#17116}
2017-03-08 13:42:26 +00:00
zstein
d5a2d9ad0c WebRtcVideoChannel2Test::SetRecvCodecsAcceptsMultipleVideoCodecs passes now.
WebRtcVideoChannel2Test::SetRecvCodecsSetsFecForAllVideoCodecs was never
fully implemented and hasn't been touched in over a year.

BUG=NONE

Review-Url: https://codereview.webrtc.org/2736483002
Cr-Commit-Position: refs/heads/master@{#17075}
2017-03-06 20:09:24 +00:00
tommi
db23ea69b6 Add performance tracing for PlatformThread and parts of the video code.
BUG=webrtc:7219

Review-Url: https://codereview.webrtc.org/2729783004
Cr-Commit-Position: refs/heads/master@{#17009}
2017-03-03 15:21:18 +00:00
tommi
f0d7b2b507 Make sure that ApmHelpersTest tests don't enter a tight loop.
BUG=webrtc:7219, webrtc:7290
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2729133002
Cr-Commit-Position: refs/heads/master@{#16996}
2017-03-03 09:34:15 +00:00
ossu
0c4b849b27 Pick a matching CN codec, rather than the first CN codec.
It seems to me that we're currently just picking the first CN codec, rather than the one that matches the clock rate of the voice codec. The only test I've gotten to fail by changing this behavior is the one that's also changed in this CL, which explicitly expects a CN codec to be chosen even though there's none matching.

BUG=webrtc:7282

Review-Url: https://codereview.webrtc.org/2707133007
Cr-Commit-Position: refs/heads/master@{#16979}
2017-03-02 19:03:25 +00:00
solenberg
796b8f9d71 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
2017-03-02 01:02:23 +00:00
solenberg
2100c0ba13 Support N unsignaled audio streams.
BUG=webrtc:7175, webrtc:5208

Review-Url: https://codereview.webrtc.org/2685893002
Cr-Commit-Position: refs/heads/master@{#16952}
2017-03-01 19:29:29 +00:00
sprang
f31969a584 Disable flaky test while working on improvement.
WebRtcVideoEngine2Test.RecreatesEncoderOnContentTypeChange is flaky.
Disable it temporarily, while working on improvement.

TBR=pthatcher@webrtc.org
BUG=webrtc:7275

Review-Url: https://codereview.webrtc.org/2727503003
Cr-Commit-Position: refs/heads/master@{#16937}
2017-03-01 10:27:34 +00:00
brandtr
340e3fd59f Split FlexFEC field trial in two.
- The "flexfec-03" codec is advertised in the SDP whenever the
  "WebRTC-FlexFEC-03-Advertised" field trial is enabled.
- Sending FlexFEC packets is enabled whenever the "flexfec-03" codec is
  negotiated, and the "WebRTC-FlexFEC-03" field trial is enabled.

After this CL, the number of calls to
WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec during renegotiation
will be reduced for cases when only one endpoint has the "WebRTC-FlexFEC-03"
field trial enabled.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2716733005
Cr-Commit-Position: refs/heads/master@{#16925}
2017-02-28 23:43:10 +00:00
sprang
f24a06406a Recreate WebrtcVideoSendStream if screen content setting is changed.
This avoids the situation where an encoder, not supporting certain
screen content settings, is created for a config where screencast is
off, and later ReconfigureEncoder() is called updating the configuration
but not the encoder instance, causing an inconsistency in the encoder's
InitEncode() call.

TBR=pthatcher@webrtc.org
BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2710493008
Cr-Commit-Position: refs/heads/master@{#16921}
2017-02-28 21:23:26 +00:00
sprang
c1b57a15bf Test field trial group with startswith rather than equals.
BUG=webrtc:7266

Review-Url: https://codereview.webrtc.org/2717973005
Cr-Commit-Position: refs/heads/master@{#16915}
2017-02-28 16:50:47 +00:00
tommi
322a9e42f8 Handle TimeUntilNextProcess in StartupShutdownWithExternalADM
BUG=webrtc:7258

Review-Url: https://codereview.webrtc.org/2719113002
Cr-Commit-Position: refs/heads/master@{#16894}
2017-02-28 10:12:57 +00:00
solenberg
76377c55b7 Remove usage of VoEAudioProcessing from WVoE/MC.
Calling APM and TransmitMixer directly instead.

BUG=webrtc:4690
TBR=peah@webrtc.org

Review-Url: https://codereview.webrtc.org/2681033010
Cr-Commit-Position: refs/heads/master@{#16734}
2017-02-21 08:54:31 +00:00
nisse
1e32122168 Delete VideoCaptureCapability::codecType and related logic.
The video_capture module includes remnants of support for cameras
producing encoded frames. However, this seems to be unused, and is
explicitly not supported by VideoCaptureImpl::IncomingFrame.

BUG=None

Review-Url: https://codereview.webrtc.org/2668693008
Cr-Commit-Position: refs/heads/master@{#16732}
2017-02-21 07:27:37 +00:00
mzanaty
8a855d6916 Allow any unsignalled SSRC changes on default video receive channel.
The first unsignalled SSRC creates a default receive channel.
Any unsignalled SSRC changes after that replace the default SSRC.
Add unit tests for changing unsignalled SSRCs.

BUG=webrtc:5208

Review-Url: https://codereview.webrtc.org/2692993009
Cr-Commit-Position: refs/heads/master@{#16682}
2017-02-17 23:46:43 +00:00
solenberg
4904fb6f46 Be less pessimistic about turning "default" receive streams into signaled streams.
BUG=webrtc:7179, b/34746131

Review-Url: https://codereview.webrtc.org/2685573003
Cr-Commit-Position: refs/heads/master@{#16673}
2017-02-17 20:01:14 +00:00
nisse
642943baea Delete DeviceInfoImpl::GetExpectedCaptureDelay and related declarations.
This feature is unused. We can then also delete the header file
video_capture_delay.h.

BUG=None

Review-Url: https://codereview.webrtc.org/2665113006
Cr-Commit-Position: refs/heads/master@{#16666}
2017-02-17 14:22:07 +00:00
sprang
3ebabf1c29 Screen content simulcast layers should not be downscaled.
Fix config so, size isn't downscaled, add unit test coverage.

BUG=webrtc:7171, webrtc:4172

Review-Url: https://codereview.webrtc.org/2692343007
Cr-Commit-Position: refs/heads/master@{#16648}
2017-02-16 15:35:22 +00:00
ossu
11bfc53cd9 Fixed a couple of build-flag dependent tests of webrtcvoiceengine.
The codecs expected by HasCorrectCodecs now depends which codecs were
enabled by build flags.

SendSideBweWithOverheadTest.MinAndMaxBitrate now expects different
values for min bitrate depending on if we support 120 ms frames for
Opus.

BUG=b/35415435

Review-Url: https://codereview.webrtc.org/2691343008
Cr-Commit-Position: refs/heads/master@{#16643}
2017-02-16 13:37:06 +00:00
Peter Boström
f812e45d8f Handle InitDecode and reset in fallback decoder.
Makes sure video decoder software fallback handles InitDecode()
failures, and properly releases the pointer after ::Release() so that
another decode failure will properly reinitialize the decoder.

Also makes sure to not call Decode() without a previous InitDecode()
succeeding.

BUG=webrtc:7154
R=noahric@chromium.org, sophiechang@chromium.org

Review-Url: https://codereview.webrtc.org/2690183004 .
Cr-Commit-Position: refs/heads/master@{#16594}
2017-02-13 22:11:08 +00:00
ossu
bcd88dbc01 WebRtcVoiceEngineTest: Changed a static_cast to a checked_cast.
Also two spelling fixes.
This is a follow-up to https://codereview.webrtc.org/2669583002/

TBR=kwiberg@webrtc.org
BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2697453004
Cr-Commit-Position: refs/heads/master@{#16586}
2017-02-13 15:04:05 +00:00
deadbeef
112b2e99d8 Switching some interfaces to use std::unique_ptr<>.
This helps show where ownership is transfered between objects.

Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.

BUG=None
TBR=magjed@webrtc.org

Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
2017-02-11 04:13:37 +00:00
kwiberg
087bd34d23 Move AudioDecoder and related stuff to the api/ directory
BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
2017-02-10 16:15:44 +00:00
sprang
84a3759825 Change rtc::VideoSinkWants to have target and a max pixel count
The current method with max_pixel_count and max_pixel_count_step_up,
where only one should be used at a time and this first signaling an
inclusive upper bound and other other an exclusive lower bound, makes
for a lot of confusion.

I've updated this to have a desired target and a maximum instead. The
source should select a resolution as close to the target as possible,
but no higher than the maximum.

I intend to also add similar frame rate settings in an upcoming cl.

BUG=webrtc:4172,webrtc:6850

Review-Url: https://codereview.webrtc.org/2672793002
Cr-Commit-Position: refs/heads/master@{#16533}
2017-02-10 15:04:27 +00:00
ivoc
4ca18696c5 Allow residual echo detector to be enabled/disabled using AudioOptions, and no longer disable it on mobile platforms.
BUG=webrtc:7136

Review-Url: https://codereview.webrtc.org/2679103007
Cr-Commit-Position: refs/heads/master@{#16531}
2017-02-10 13:11:09 +00:00
solenberg
55c5be0a03 Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel.
These methods relate to typing detection but are not used anymore. Typing detection is enabled through the VoiceDetection module on the APM.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2684933008
Cr-Commit-Position: refs/heads/master@{#16527}
2017-02-10 09:20:25 +00:00