278 Commits

Author SHA1 Message Date
michaelt
79e05888e8 Set actual transport overhead in rtp_rtcp
BUG=webrtc:6557

Review-Url: https://codereview.webrtc.org/2437503004
Cr-Commit-Position: refs/heads/master@{#14968}
2016-11-08 10:50:16 +00:00
magjed
23b7a4a390 Refactor WebRtcVideoReceiveStream::FilterSupportedCodecs
This CL removes WebRtcVideoReceiveStream::FilterSupportedCodecs and
adds more fine grained code for selecting and verifying codecs. This
also removes unnecessary copying of cricket::VideoCodecs.

BUG=webrtc:6337

Review-Url: https://codereview.webrtc.org/2474433012
Cr-Commit-Position: refs/heads/master@{#14964}
2016-11-08 09:13:02 +00:00
magjed
3663c52382 Provide move semantic for cricket::Codec and subclasses
The cricket::Codec class contains std containers like
std::map<std::string, std::string> and is expensive to copy. This CL
adds move constructors and move assignment operators for it and all
subclasses.

This CL also:
 * Implement functions with '= default' instead of doing it manually.
 * Makes codec::Matches symmetric. We currently don't check if the
   payload type of the callee is static, and might incorrectly return
   true in these cases.

BUG=None

Review-Url: https://codereview.webrtc.org/2481193002
Cr-Commit-Position: refs/heads/master@{#14956}
2016-11-07 18:14:44 +00:00
minyue
10cbb4648f Fixing config for Audio BWE.
The unit was kbps but the one default use of it is in bps. The inconsistency should be fixed.

BUG=webrtc:6670

Review-Url: https://codereview.webrtc.org/2247213005
Cr-Commit-Position: refs/heads/master@{#14955}
2016-11-07 17:29:27 +00:00
brandtr
87d7d77700 Add new codec for FlexFEC.
This CL does nothing except adding new strings and enums corresponding to
the new codec.

R=stefan@webrtc.org
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2470103002
Cr-Commit-Position: refs/heads/master@{#14943}
2016-11-07 11:04:03 +00:00
kwiberg
37b8b11661 Revert of Removed the legacy behavior of stopping playout when setting new receive codecs. (patchset #1 id:1 of https://codereview.webrtc.org/2409483003/ )
Reason for revert:
Reverting because of the reasons given in comment #16:

"This change breaks a scenario that is unfortunately not covered by unit tests,
but can easily happen in a real call.

The scenario that is broken by the change is this:
1. A sends an offer to B, with a set of codecs C_a (which is a subset of C_b,
the codecs supported by B)
2. B responds with an answer, and sets the receive codecs to C_a.
3. At a later time, B generates a new offer which by default includes all codecs
in C_b.
4. B calls SetLocalDescription() with this offer, that adds new receive codecs.
5. Adding the new codecs fails, because of the check at
https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/channel.....
This causes SetLocalDescription() itself to fail. The net effect is that B
cannot set a local description it just generated.

Before the CL mentioned above, we'd stop playout before changing the codecs, and
the operation would succeed."

Original issue's description:
> Removed the legacy behavior of stopping playout when setting new receive codecs.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/917d4e1e7131f35764cff932a8793151585e8179
> Cr-Commit-Position: refs/heads/master@{#14610}

TBR=solenberg@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2478433003
Cr-Commit-Position: refs/heads/master@{#14905}
2016-11-03 09:47:02 +00:00
magjed
d2fce1744f Suppress WebRtcVideoEncoderFactory overloaded virtual function warning
Suppress WebRtcVideoEncoderFactory overloaded virtual function warning
in WebRtcSimulcastEncoderFactory and FakeWebRtcVideoEncoderFactory.

This warning is triggered by the change in this CL:
https://codereview.webrtc.org/2449993003/.

BUG=webrtc:6402, webrtc:6337

Review-Url: https://codereview.webrtc.org/2468253002
Cr-Commit-Position: refs/heads/master@{#14901}
2016-11-02 18:08:37 +00:00
nisse
7341ab8e25 Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
Reason for revert:
Breaks chrome, see https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/19019/steps/compile/logs/stdio

Analysis: Chrome uses cricket::VideoFrame, without explicitly including webrtc/media/base/videoframe.h, and breaks when that file is no longer included by any other webrtc headers. Will reland after updating Chrome.

Original issue's description:
> Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
>
> Replaced with webrtc::VideoFrame.
>
> TBR=mflodman@webrtc.org
> BUG=webrtc:5682
>
> Committed: https://crrev.com/45c8b8940042bd2574c39920804ade8343cefdba
> Cr-Commit-Position: refs/heads/master@{#14885}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,tkchin@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2471783002
Cr-Commit-Position: refs/heads/master@{#14886}
2016-11-02 10:40:05 +00:00
nisse
45c8b89400 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame.
Replaced with webrtc::VideoFrame.

TBR=mflodman@webrtc.org
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2383093002
Cr-Commit-Position: refs/heads/master@{#14885}
2016-11-02 10:20:28 +00:00
perkj
803d97f159 Let ViEEncoder express resolution requests as Sinkwants.
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.

To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.

BUG=webrtc:5687
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
2016-11-01 18:45:54 +00:00
stefan
b521aa704f Clean up abs-send-time for audio.
BUG=None

Review-Url: https://codereview.webrtc.org/2455013003
Cr-Commit-Position: refs/heads/master@{#14870}
2016-11-01 10:17:18 +00:00
nisse
67dca9f12e Delete ShallowCopy, in favor of copy construction and assignment.
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2443123002
Cr-Commit-Position: refs/heads/master@{#14853}
2016-10-31 15:05:58 +00:00
sakal
87da404883 Implement qpSum stat for video send ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2430603003
Cr-Commit-Position: refs/heads/master@{#14851}
2016-10-31 13:53:51 +00:00
minyue
6b825df37e Using AudioOption to enable audio network adaptor.
BUG=webrtc:6303

Review-Url: https://codereview.webrtc.org/2397573006
Cr-Commit-Position: refs/heads/master@{#14845}
2016-10-31 11:08:37 +00:00
Per
21d45d2ab6 Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
This is the second reland.  Patchset 1 contains the reverted cl.
Patchset 2 revert the change to initialize the encoder with resolution 1*1pixels if an internal source is used.
This is to to fix the problem reported in https://codereview.webrtc.org/2457203002/ https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/35251 remoting.
Fix has been verified to work in Chrome.
This reverts commit 05a55b500d83e4212d4e54f0fecf13097e782ffa.

BUG=webrtc:6371 b/32285861
TBR=pbos@webrtc.org, skvlad@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2458363002 .

Cr-Commit-Position: refs/heads/master@{#14833}
2016-10-30 20:38:56 +00:00
emircan
05a55b500d Revert of Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #2 id:20001 of https://codereview.webrtc.org/2455963004/ )
Reason for revert:
It breaks webrtc.fyi bots, see
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/35251.

Original issue's description:
> Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
>
> Patchset 1 contain the originally reviewed cl in https://codereview.webrtc.org/2455063002/
> TBR=stefan@webrtc.org, pbos@webrtc.org, skvlad@webrtc.org
>
> BUG=webrtc:6371 b/32285861
>
> Committed: https://crrev.com/5f1b05129e4770c98429164761779d99a410e7c8
> Cr-Commit-Position: refs/heads/master@{#14823}

TBR=pbos@webrtc.org,skvlad@webrtc.org,stefan@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6371 b/32285861

Review-Url: https://codereview.webrtc.org/2457203002
Cr-Commit-Position: refs/heads/master@{#14829}
2016-10-28 21:06:36 +00:00
magjed
1e45cc6ee0 Replace WebRtcVideoEncoderFactory::VideoCodec with cricket::VideoCodec
This CL introduces two new functions to the WebRtcVideoEncoderFactory
interface based on cricket::VideoFormat instead of
WebRtcVideoEncoderFactory::VideoCodec. The functions are:
WebRtcVideoEncoderFactory::CreateVideoEncoder() and
WebRtcVideoEncoderFactory::supported_codecs(). In order to make a smooth
transition to the new interface, the old functions are kept, and default
implementations are provided for both the old and new functions so that
external clients can switch from the old to the new functions in peace.
The default implementations will just convert between
cricket::VideoFormat and WebRtcVideoEncoderFactory::VideoCodec. Once all
external clients have updated their code, the plan is to remove the old
functions and all default implementations to make
WebRtcVideoEncoderFactory a pure interface again.

BUG=webrtc:6402,webrtc:6337

Review-Url: https://codereview.webrtc.org/2449993003
Cr-Commit-Position: refs/heads/master@{#14826}
2016-10-28 14:43:52 +00:00
perkj
5f1b05129e Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
Patchset 1 contain the originally reviewed cl in https://codereview.webrtc.org/2455063002/
TBR=stefan@webrtc.org, pbos@webrtc.org, skvlad@webrtc.org

BUG=webrtc:6371 b/32285861

Review-Url: https://codereview.webrtc.org/2455963004
Cr-Commit-Position: refs/heads/master@{#14823}
2016-10-28 13:58:43 +00:00
perkj
5da65f241c Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ )
Reason for revert:
Seems to break WebRTC perf tests.

Original issue's description:
> Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
>
> BUG=b/32285861
>
> Committed: https://crrev.com/461c29e436b5bd7ed019e83024e24dc8e86ec9b9
> Cr-Commit-Position: refs/heads/master@{#14813}

TBR=skvlad@webrtc.org,pbos@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=b/32285861

Review-Url: https://codereview.webrtc.org/2457083002
Cr-Commit-Position: refs/heads/master@{#14815}
2016-10-28 11:48:53 +00:00
perkj
461c29e436 Change ViEEncoder to not reconfigure the encoder until the video resolution is known.
BUG=b/32285861

Review-Url: https://codereview.webrtc.org/2455063002
Cr-Commit-Position: refs/heads/master@{#14813}
2016-10-28 10:13:58 +00:00
sakal
e5ba44eab1 Implement framesDecoded stat in video receive ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2423823003
Cr-Commit-Position: refs/heads/master@{#14789}
2016-10-26 14:09:29 +00:00
solenberg
059fb4480b - Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing.
- Update MockAudioProcessing to current APM interface.
- Replace calls to VoEAudioProcessing::Start/StopAecDump with direct calls on APM.
- Add AudioProcessing* in WVoE, get it from VoE, so we can call directly on APM.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2446143002
Cr-Commit-Position: refs/heads/master@{#14786}
2016-10-26 12:12:29 +00:00
solenberg
88b7074745 Remove unused function implementations from FakeWebRtcVoiceEngine.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2440353003
Cr-Commit-Position: refs/heads/master@{#14752}
2016-10-24 20:46:04 +00:00
sakal
43536c3d6a Implement framesEncoded stat in video send ssrc stats.
Implemented as defined by this pull request: https://github.com/w3c/webrtc-stats/pull/70

BUG=webrtc:6541

Review-Url: https://codereview.webrtc.org/2421193003
Cr-Commit-Position: refs/heads/master@{#14734}
2016-10-24 09:09:39 +00:00
perkj
267527459b Remove cricket::VideoCodec with, height and framerate properties
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.

This cl require some upstream projects to change first.

BUG=webrtc:5332

Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
2016-10-24 08:21:24 +00:00
brandtr
25445d3d4b Integrate FlexfecReceiveStream with Call.
Call demultiplexes received RTP packets and delivers these to the
appropriate {Video,Flexfec}ReceiveStreams. A single video stream
could conceivably be protected by multiple FlexFEC streams.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2388303009
Cr-Commit-Position: refs/heads/master@{#14727}
2016-10-24 06:37:24 +00:00
ivoc
8c63a82bf5 Add a placeholder stat for logging the estimated residual echo likelihood.
The stat is currently always set to zero until the residual echo detector has landed.

BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2431443003
Cr-Commit-Position: refs/heads/master@{#14721}
2016-10-21 11:10:08 +00:00
minyue
7a973447eb Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
BUG=webrtc:5806, webrtc:4690

Review-Url: https://codereview.webrtc.org/2405183002
Cr-Commit-Position: refs/heads/master@{#14700}
2016-10-20 10:27:21 +00:00
aleloi
e33c5d918a Added a level controller initialization value to MediaConstraints.
An audio track with a level controller with the correct initialization
value can be created by a combination of
PeerConnectionFactory::CreateAudioTrack(..., audio_source) and
either
audio_source = PeerConnectionFactory::CreateAudioSource(constraints) or
audio_source = PeerConnectionFactory::CreateAudioSource(audio_options).

NOTRY=True
BUG=webrtc:6386

Review-Url: https://codereview.webrtc.org/2408143003
Cr-Commit-Position: refs/heads/master@{#14693}
2016-10-20 08:53:30 +00:00
Per
061ea0df03 Remove VideoCodec resolution validation.
This is needed to be able to land https://codereview.webrtc.org/2408153002/  "Remove cricket::VideoCodec with, height and framerate properties" without breaking upstream projects.

BUG=webrtc:5332
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2437453003 .

Cr-Commit-Position: refs/heads/master@{#14681}
2016-10-19 11:58:09 +00:00
nisse
09347858f7 Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ )
This cl now makes cricket::VideoFrame and cricket::WebRtcVideoFrame aliases for webrtc::VideoFrame.

Reason for revert:
Fixing backwards compatibility issues.

Original issue's description:
> Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
>
> Reason for revert:
> Breaks compile for Chromium builds:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142
>
> FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
> ../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
>                  weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
>                                              ~~~~~ ^
> 1 error generated.
>
> Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.
>
> Original issue's description:
> > Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> > all methods but a few constructors. And similarly for the
> > subclass cricket::WebRtcVideoFrame.
> >
> > TBR=tkchin@webrtc.org  # Added an include line
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> > Cr-Commit-Position: refs/heads/master@{#14576}
>
> TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
> NOTRY=True
> NOPRESUBMIT=True
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d36dd499c8f253cbcf37364c2a070c2e8c7100e9
> Cr-Commit-Position: refs/heads/master@{#14583}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2411953002
Cr-Commit-Position: refs/heads/master@{#14678}
2016-10-19 07:30:35 +00:00
michaelt
53fe19d6f3 Set min and max rate on caller and on callee side.
BUG=webrtc:6518

Review-Url: https://codereview.webrtc.org/2410903002
Cr-Commit-Position: refs/heads/master@{#14666}
2016-10-18 16:39:28 +00:00
nisse
e5ddf52b97 Delete unused file webrtcvideochannelfactory.h.
BUG=None

Review-Url: https://codereview.webrtc.org/2416453002
Cr-Commit-Position: refs/heads/master@{#14641}
2016-10-15 18:29:59 +00:00
solenberg
917d4e1e71 Removed the legacy behavior of stopping playout when setting new receive codecs.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2409483003
Cr-Commit-Position: refs/heads/master@{#14610}
2016-10-12 10:20:34 +00:00
peah
73a28ee066 The AudioProcessing class is used as an interface
to the functionality in the audio processing module.
Therefore, it should be a pure interface.
This CL ensures that is the case.

BUG=webrtc:6515

Review-Url: https://codereview.webrtc.org/2406193002
Cr-Commit-Position: refs/heads/master@{#14608}
2016-10-12 10:01:57 +00:00
kjellander
d36dd499c8 Revert of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #9 id:160001 of https://codereview.webrtc.org/2315663002/ )
Reason for revert:
Breaks compile for Chromium builds:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/10761
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/18142

FAILED: obj/remoting/protocol/protocol/webrtc_video_renderer_adapter.o
../../remoting/protocol/webrtc_video_renderer_adapter.cc:110:52: error: no member named 'transport_frame_id' in 'cricket::VideoFrame'
                 weak_factory_.GetWeakPtr(), frame.transport_frame_id(),
                                             ~~~~~ ^
1 error generated.

Please run chromium trybots as described at https://webrtc.org/contributing/#tryjobs-on-chromium-trybots before relanding.

Original issue's description:
> Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
> all methods but a few constructors. And similarly for the
> subclass cricket::WebRtcVideoFrame.
>
> TBR=tkchin@webrtc.org  # Added an include line
> BUG=webrtc:5682
>
> Committed: https://crrev.com/dda6ec008a0fc8d52e118814fb779032e8931968
> Cr-Commit-Position: refs/heads/master@{#14576}

TBR=perkj@webrtc.org,pthatcher@webrtc.org,pthatcher@chromium.org,tkchin@webrtc.org,nisse@webrtc.org
NOTRY=True
NOPRESUBMIT=True
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2402853002
Cr-Commit-Position: refs/heads/master@{#14583}
2016-10-09 05:21:40 +00:00
skvlad
11a9cbfa50 Refactoring: move ownership of RtcEventLog from Call to PeerConnection
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.

This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).

BUG=webrtc:6393

Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
2016-10-07 18:53:15 +00:00
nisse
dda6ec008a Make cricket::VideoFrame inherit webrtc::VideoFrame. Delete
all methods but a few constructors. And similarly for the
subclass cricket::WebRtcVideoFrame.

TBR=tkchin@webrtc.org  # Added an include line
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2315663002
Cr-Commit-Position: refs/heads/master@{#14576}
2016-10-07 15:16:59 +00:00
brandtr
b5f2c3fbe9 Rename FecConfig to UlpfecConfig in config.h.
Also rename some related minor methods. No functional changes
are intended/expected.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2391963002
Cr-Commit-Position: refs/heads/master@{#14513}
2016-10-05 06:28:43 +00:00
aleloi
18e0b67815 Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
BUG=webrtc:6346

Review-Url: https://codereview.webrtc.org/2383143002
Cr-Commit-Position: refs/heads/master@{#14493}
2016-10-04 09:45:54 +00:00
perkj
fa10b557d9 Releand of Let ViEEncoder handle resolution changes.
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.

Original cl description:
Let ViEEncoder handle resolution changes.

This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
2016-10-03 06:45:33 +00:00
kwiberg
ac9f876bc0 Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
gmock.h and gtest.h were moved (or rather, got wrappers so that we
could put some icky compatibility hacks in one place instead of 500)
in this CL: https://codereview.webrtc.org/2358993004/

NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2381013002
Cr-Commit-Position: refs/heads/master@{#14464}
2016-10-01 05:29:53 +00:00
nisse
f5297a019e Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ )
Reason for revert:
Downstream code is being fixed.

Original issue's description:
> Revert of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #9 id:160001 of https://codereview.webrtc.org/2262443003/ )
>
> Reason for revert:
> Breaks downstream testcode, still using CapturedFrame.
>
> Original issue's description:
> > Delete VideoFrameFactory, CapturedFrame, and related code.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/66ac50e58c790624d51ede10ae438cbadbca9d2e
> > Cr-Commit-Position: refs/heads/master@{#14315}
>
> TBR=pthatcher@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/66492210e57ee8efce2ad4d45a8781df1fcaa5e3
> Cr-Commit-Position: refs/heads/master@{#14320}

TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2370993003
Cr-Commit-Position: refs/heads/master@{#14453}
2016-09-30 08:34:35 +00:00
perkj
3b703ede8b Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
Reason for revert:
Fails on a content_browsertest (and also webrtc_perf?)

https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Tester/builds/34336

https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/9091/steps/webrtc_perf_tests/logs/stdio
[  FAILED  ] FullStackTest.ParisQcifWithoutPacketLoss (59436 ms)

Original issue's description:
> Let ViEEncoder handle resolution changes.
>
> This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
>
> With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
>
> BUG=webrtc:5687, webrtc:6371, webrtc:5332
>
> Committed: https://crrev.com/26105b41b4f97642ee30cb067dc786c2737709ad
> Cr-Commit-Position: refs/heads/master@{#14445}

TBR=sprang@webrtc.org,mflodman@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2383493005
Cr-Commit-Position: refs/heads/master@{#14447}
2016-09-30 06:25:46 +00:00
perkj
26105b41b4 Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
2016-09-30 05:39:15 +00:00
Per
a48ddb7636 Add VideoSendStream::Stats::prefered_media_bitrate_bps
This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured.
This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change
from WebRtcVideoEngine2 to ViEEncoder.

BUG=webrtc:6371
R=mflodman@webrtc.org, sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2368223002 .

Cr-Commit-Position: refs/heads/master@{#14431}
2016-09-29 09:49:01 +00:00
kwiberg
77eab70470 Enable the -Wundef warning for clang
NOPRESUBMIT=true
BUG=webrtc:6398

Review-Url: https://codereview.webrtc.org/2358993004
Cr-Commit-Position: refs/heads/master@{#14425}
2016-09-29 00:42:08 +00:00
palmkvist
e75f204b06 Expose Ivf logging through the native API
BUG=webrtc:6300

Review-Url: https://codereview.webrtc.org/2303273002
Cr-Commit-Position: refs/heads/master@{#14419}
2016-09-28 13:19:53 +00:00
nisse
e5684c5387 Delete method webrtc::VideoFrame::allocated_size and enum PlaneType.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2380623002
Cr-Commit-Position: refs/heads/master@{#14416}
2016-09-28 10:14:15 +00:00
kthelgason
29a44e351e This is a resubmission of https://codereview.webrtc.org/2047513002/
Original description:
Add proper lifetime of encoder-specific settings.

Permits passing VideoEncoderConfig between threads and not worry about
the lifetime of an underlying void pointer. Also adds type safety to
unpacking of codec-specific settings.

These settings are not yet propagating to VideoEncoder interfaces, but
the aim is to get rid of webrtc::VideoCodec for VideoEncoder.

BUG=webrtc:3424
R=perkj@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review-Url: https://codereview.webrtc.org/2347843002
Cr-Commit-Position: refs/heads/master@{#14396}
2016-09-27 10:52:05 +00:00