14103 Commits

Author SHA1 Message Date
philipel
aee3e0eb32 Only advance |first_seq_num_| if packets are explicitly cleared from the PacketBuffer.
In this CL:
 - Don't insert a packet if we have explicitly cleared past it.
 - Added some logging to ExpandBufferSize.
 - Renamed IsContinuous to PotentialNewFrame.
 - Unittests updated/added for this new behavior.
 - Refactored TestPacketBuffer unittests.

BUG=webrtc:5514
R=danilchap@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2399373002 .

Cr-Commit-Position: refs/heads/master@{#14871}
2016-11-01 10:45:43 +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
charujain
aca3a249c3 Moving stun_prober target from webrtc/p2p to webrtc/examples
BUG=webrtc:6440
NOTRY=True

Review-Url: https://codereview.webrtc.org/2460343002
Cr-Commit-Position: refs/heads/master@{#14869}
2016-11-01 10:09:19 +00:00
hbos
eeafe94f28 RTCInboundRTPStreamStats[1] added.
Not all stats are collected in this CL, this must be addressed before
closing the issue.

[1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*

  Re-landed after having to be reverted
  https://codereview.webrtc.org/2470683002/ due to depending on a CL
  that was reverted. Now that that has re-landed
  https://codereview.webrtc.org/2470703002/ this is ready to re-land.

BUG=chromium:627816, chromium:657855, chromium:657854
R=hta@webrtc.org
TBR=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/2465173003
Cr-Commit-Position: refs/heads/master@{#14868}
2016-11-01 10:00:24 +00:00
sprang
b84ad63b0a Add RTCP packet class for signaling encoder target bitrate.
This is a proposal for a new RTCP message. Feel free to comment on the
message structure, selected type ids etc, as well as code for
serialization/deserialization. Once we agree on this, I'll continue
with wiring it up in the actual rtcp sender and receiver.

BUG=webrtc:6301

Review-Url: https://codereview.webrtc.org/2306873003
Cr-Commit-Position: refs/heads/master@{#14867}
2016-11-01 09:50:17 +00:00
hbos
6ded190864 RTCOutboundRTPStreamStats[1] added.
This also adds RTCRTPStreamStats[2] which it derives from. Not all stats
are supported in this CL, this must be addressed before closing the
issue.

RTCStatsReport also gets a timestamp and ToString.

[1] https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
[2] https://w3c.github.io/webrtc-stats/#streamstats-dict*

  This was previously reverted https://codereview.webrtc.org/2465223002/
  because RTCStatsReport::Create added a new parameter not used by
  Chromium unittests. Temporarily added a default value to the argument
  to be removed after rolling and updating Chromium.

BUG=chromium:627816, chromium:657856, chromium:657854
TBR=deadbeef@webrtc.org

Review-Url: https://codereview.webrtc.org/2470703002
Cr-Commit-Position: refs/heads/master@{#14866}
2016-11-01 08:50:52 +00:00
johan
15ca8f6aeb Let receiving() and SignalRecevingState be part of rtc::PacketTransportInterface.
Writable() and the related signal are already part of rtc::PacketTransportInterface. Sense of code symmetry aesthetics dictates that receiving() and the related signal should be declared in the same place.

BUG=webrtc:6531

Review-Url: https://codereview.webrtc.org/2444793003
Cr-Commit-Position: refs/heads/master@{#14865}
2016-11-01 08:47:48 +00:00
asapersson
fe647f4ab2 Add ability to handle data from multiple streams in RateAccCounter.
BUG=webrtc:5283

Review-Url: https://codereview.webrtc.org/2235223002
Cr-Commit-Position: refs/heads/master@{#14864}
2016-11-01 07:21:41 +00:00
perkj
7eaa83622b Revert of RTCOutboundRTPStreamStats added. (patchset #3 id:80001 of https://codereview.webrtc.org/2456463002/ )
Reason for revert:
Breaks Chrome FYI.
peerconnection_unittest calls RTCStatsReport::Create without  parameters.

Original issue's description:
> RTCOutboundRTPStreamStats[1] added.
>
> This also adds RTCRTPStreamStats[2] which it derives from. Not all stats
> are supported in this CL, this must be addressed before closing the
> issue.
>
> RTCStatsReport also gets a timestamp and ToString.
>
> [1] https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
> [2] https://w3c.github.io/webrtc-stats/#streamstats-dict*
>
> BUG=chromium:627816, chromium:657856, chromium:657854
>
> Committed: https://crrev.com/69e9cb08285f6cbcab547c7a5e6aa668fa6f2d29
> Cr-Commit-Position: refs/heads/master@{#14860}

TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:627816, chromium:657856, chromium:657854

Review-Url: https://codereview.webrtc.org/2465223002
Cr-Commit-Position: refs/heads/master@{#14863}
2016-11-01 06:52:28 +00:00
perkj
4ed075034a Revert of RTCInboundRTPStreamStats added. (patchset #4 id:100001 of https://codereview.webrtc.org/2452043002/ )
Reason for revert:
Dependend cl Breaks Chrome FYI.
peerconnection_unittest anropar RTCStatsReport::Create without  parameters.

Original issue's description:
> RTCInboundRTPStreamStats[1] added.
>
> Not all stats are collected in this CL, this must be addressed before
> closing the issue.
>
> [1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
>
> BUG=chromium:627816, chromium:657855, chromium:657854
>
> Committed: https://crrev.com/0d7bf169402ea9345d163998f4f7df89229ac470
> Cr-Commit-Position: refs/heads/master@{#14861}

TBR=hta@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:627816, chromium:657855, chromium:657854

Review-Url: https://codereview.webrtc.org/2470683002
Cr-Commit-Position: refs/heads/master@{#14862}
2016-11-01 06:51:00 +00:00
hbos
0d7bf16940 RTCInboundRTPStreamStats[1] added.
Not all stats are collected in this CL, this must be addressed before
closing the issue.

[1] https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*

BUG=chromium:627816, chromium:657855, chromium:657854

Review-Url: https://codereview.webrtc.org/2452043002
Cr-Commit-Position: refs/heads/master@{#14861}
2016-10-31 22:31:09 +00:00
hbos
69e9cb0828 RTCOutboundRTPStreamStats[1] added.
This also adds RTCRTPStreamStats[2] which it derives from. Not all stats
are supported in this CL, this must be addressed before closing the
issue.

RTCStatsReport also gets a timestamp and ToString.

[1] https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
[2] https://w3c.github.io/webrtc-stats/#streamstats-dict*

BUG=chromium:627816, chromium:657856, chromium:657854

Review-Url: https://codereview.webrtc.org/2456463002
Cr-Commit-Position: refs/heads/master@{#14860}
2016-10-31 21:48:44 +00:00
Henrik Kjellander
bb9212a33e Add ffmpeg and zxing to webrtc/tools/video_quality_toolchain.
Usually .sha1 files are downlaoded using DEPS hooks but since this
bucket is internal we can't run it everywhere since it would fail
non-Googler checkouts. Instead we download the binaries by calling
a Python script, which will be added as a separate build step on the
buildbots.

The .sha1 files are copied from
https://cs.chromium.org/chromium/src/chrome/test/data/webrtc/resources/tools/
leaving out pesq and sox.

BUG=webrtc:6633
TESTED=Ran the download.py script on Mac and verified the files were downloaded.
R=mandermo@google.com, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/2462023002 .

Cr-Commit-Position: refs/heads/master@{#14859}
2016-10-31 21:02:36 +00:00
solenberg
e566ac7341 Remove voe::Channel::StopReceive() and associated logic.
- The legacy API is not used in WVoE/MC.
- Removed use of the API (along with StartReceive()) from unit tests.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2453243003
Cr-Commit-Position: refs/heads/master@{#14858}
2016-10-31 19:52:39 +00:00
magjed
9c41e47b12 Remove unnecessary test fixture in codec_unittest.cc
BUG=None

Review-Url: https://codereview.webrtc.org/2462053002
Cr-Commit-Position: refs/heads/master@{#14857}
2016-10-31 16:06:07 +00:00
ossu
6b6c88f184 NetEq jitter calculation now done in uint64_t.
The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.

BUG=chromium:653268

Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
2016-10-31 15:59:34 +00:00
danilchap
80ac24dd36 Allow max 1 block per type in RTCP Extended Reports
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
2016-10-31 15:40:55 +00:00
henrika
ba156cfe96 Improvements in how WebRTC.Audio.RecordedOnlyZeros is added as histogram.
Contains fixes for a non-perfect implementation in https://codereview.webrtc.org/2328433003/

Summary:

Adds WebRTC.Audio.RecordedOnlyZeros UMA stat when recording stops if:
- All level estimates during the audio session were zero, and
- If the audio session was longer than 10 seconds.

Adds four simple methods to the AudioDeviceBuffer (ADB) class to allow the ADM
to update the ADB about when media starts and stops in both directions.

Moves any "critical" parst out frome the timer (based on task queue) and ensures
that it only does trivial logging tasks.

The task queue is now owned by a unique pointer to improve control of when it
starts and stops.

Adds time measurements (for logging) of both total time playing out and total
recording time. Units are in milliseconds.

BUG=webrtc:6592

Review-Url: https://codereview.webrtc.org/2445363003
Cr-Commit-Position: refs/heads/master@{#14854}
2016-10-31 15:18:54 +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
nisse
c846f2f4c0 Fix out_frame argument of PreprocessFrameAndVerify.
Probably broken since https://codereview.webrtc.org/1482913003, making VideoProcessingTest.Resampler skip the PSNR checks.

BUG=webrtc:5259

Review-Url: https://codereview.webrtc.org/2448053003
Cr-Commit-Position: refs/heads/master@{#14852}
2016-10-31 14:20:52 +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
magjed
fffc1e5578 Add functionality for parsing H264 profile-level-id
The new code is only exercised in tests so far. The H264 profile-level-id
parsing is not complete, but it should be enough for our purposes for
now.

BUG=webrtc:6400,webrtc:6337

Review-Url: https://codereview.webrtc.org/2459633002
Cr-Commit-Position: refs/heads/master@{#14850}
2016-10-31 12:56:03 +00:00
nisse
f0a7c5ac16 Delete deprecated method VideoFrame::CreateFrame.
BUG=webrtc:6591

Review-Url: https://codereview.webrtc.org/2444383009
Cr-Commit-Position: refs/heads/master@{#14849}
2016-10-31 12:48:15 +00:00
minyue
626bc952aa Reland of "Separating video settings in VideoQualityTest".
This was landed in https://codereview.webrtc.org/2314403007/

and reverted in https://codereview.webrtc.org/2463733002/ because an error was found.

BUG=660473, webrtc:6609

Review-Url: https://codereview.webrtc.org/2466473002
Cr-Commit-Position: refs/heads/master@{#14848}
2016-10-31 12:47:09 +00:00
brandtr
869e7cd8e7 Rename ProducerFec to UlpfecGenerator.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2449783002
Cr-Commit-Position: refs/heads/master@{#14847}
2016-10-31 12:27:10 +00:00
brandtr
d55c3f68c8 Rename FecReceiver to UlpfecReceiver.
BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2451643002
Cr-Commit-Position: refs/heads/master@{#14846}
2016-10-31 11:51:38 +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
asapersson
4ee7046998 Add unit tests for bandwidth limited resolution stats in SendStatisticsProxy.
BUG=none

Review-Url: https://codereview.webrtc.org/2454343002
Cr-Commit-Position: refs/heads/master@{#14844}
2016-10-31 11:05:20 +00:00
brandtr
535830ec2d Rename Fec to Ulpfec in EndToEndTests.
This is a pure "rename CL". No functional changes are intended.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2447083002
Cr-Commit-Position: refs/heads/master@{#14843}
2016-10-31 10:46:01 +00:00
sprang
ca27f9d5b9 It seems that if encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceNum is configured to number of cores as determined by openh264 (or any number > 1 in my local tests), frame rate statistics will be mucked up (apparently thousands of frames per second) and quality will bottom out because bits per frame is then very low.
BUG=webrtc:6583

Review-Url: https://codereview.webrtc.org/2458673002
Cr-Commit-Position: refs/heads/master@{#14842}
2016-10-31 10:43:47 +00:00
brandtr
e602f0ab08 Rename Fec to Ulpfec in VideoSendStreamTest.
This is a pure "rename CL". No functional changes are intended.

BUG=webrtc:5654

Review-Url: https://codereview.webrtc.org/2449053002
Cr-Commit-Position: refs/heads/master@{#14841}
2016-10-31 10:40:56 +00:00
danilchap
42ca68ab72 Ensure one does not register same rtp header extension with different id
Added assert to RtpHeaderExtensionMap
Altered tests that did.

BUG=webrtc:1994

Review-Url: https://codereview.webrtc.org/2462663002
Cr-Commit-Position: refs/heads/master@{#14840}
2016-10-31 10:34:45 +00:00
aleloi
051f678808 Add a NeededFrequency() method to the AudioMixer::Source interface.
This change will allow for a audio source to report its sampling rate
to the audio mixer. It is needed in order to mix at a lower sampling
rate. Mixing at a lower sampling rate can in many cases lead to big
efficiency improvements, as reported by experiments.

The code affected is all implementations of the Source interface:
AudioReceiveStream and a mock class. The AudioReceiveStream now
queries its underlying voe::Channel object for the needed frequency.

Note that the changes to the mixing algorithm are done in a later CL.

BUG=webrtc:6346
NOTRY=True
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2448113009
Cr-Commit-Position: refs/heads/master@{#14839}
2016-10-31 10:26:48 +00:00
minyue
9aa78832f9 Revert of "Separating video settings in VideoQualityTest". (patchset #4 id:60001 of https://codereview.webrtc.org/2314403007/ )
Reason for revert:
Some parameters were not treated correctly. Will redo some parts.

Original issue's description:
> Reland of "Separating video settings in VideoQualityTest".
>
> Earlier trial of landing: https://codereview.webrtc.org/2312613003
>
> Reverted in https://codereview.webrtc.org/2325723002
>
> BUG=webrtc:6609
>
> Committed: https://crrev.com/16b6d6dc5b367746a9f910d1cebf9f65e8dd2c7f
> Cr-Commit-Position: refs/heads/master@{#14785}

TBR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6609

Review-Url: https://codereview.webrtc.org/2463733002
Cr-Commit-Position: refs/heads/master@{#14838}
2016-10-31 10:23:09 +00:00
kwiberg
bc8074474d Eliminate left shift of negative value by using multiplication instead
NOPRESUBMIT=true
BUG=chromium:653267

Review-Url: https://codereview.webrtc.org/2439353003
Cr-Commit-Position: refs/heads/master@{#14837}
2016-10-31 09:26:14 +00:00
tommi
c5b435dcec Re-enable PostDelayed test for TaskQueue on Windows.
The requirements have been relaxed a little bit which should allow the test to pass on our VMs that run the tests.

BUG=6610

Review-Url: https://codereview.webrtc.org/2458713005
Cr-Commit-Position: refs/heads/master@{#14836}
2016-10-31 09:17:17 +00:00
nisse
a144be384f Delete videorendererfactory.h and cricket::GdiVideoRenderer.
Ultimately, all of webrtc/media/devices should be deleted, since it is
unused in webrtc and has no unit tests. But for the time being we
need to keep cricket::GtkVideoRenderer since there are a few
applications depending on it.

BUG=webrtc:5924

Review-Url: https://codereview.webrtc.org/2460793002
Cr-Commit-Position: refs/heads/master@{#14835}
2016-10-31 07:59:54 +00:00
VladimirTechMan
18b8774792 Setting PATH so that the 'plistbuddy' utility can be found, in a typical OS X environment
Restoring a line from the older version (GYP-days) of the
build_ios_libs.sh script: modifying PATH so that the
PlistBuddy utility can be successfully found and called,
as it normally is not available under the PATH directories
in a typical OS X environment (even on developer configs).

NOTRY=True
BUG=webrtc:6372

Review-Url: https://codereview.webrtc.org/2463623002
Cr-Commit-Position: refs/heads/master@{#14834}
2016-10-31 06:09:50 +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
zijiehe
b68d655f36 Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.

After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.

BUG=webrtc:6513

Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
Review-Url: https://codereview.webrtc.org/2452263003
Cr-Original-Commit-Position: refs/heads/master@{#14830}
Cr-Commit-Position: refs/heads/master@{#14832}
2016-10-29 00:35:16 +00:00
zijiehe
fcab7d62d5 Revert of Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions (patchset #3 id:120001 of https://codereview.chromium.org/2452263003/ )
Reason for revert:
Build break in Chromium

Original issue's description:
> Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
>
> I have chosen part of 2435603010 changes to compose this change.
> According to the discussion we have made in previous change, this CL contains,
> 1. Source structure to represent a source of a DesktopCapturer.
> 2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
> DesktopCapturer.
> 3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
> DesktopCapturer APIs.
>
> After this change, We can remove WindowCapturer & ScreenCapturer references from
> Chromium, and use the new APIs.
>
> BUG=webrtc:6513
>
> Committed: https://crrev.com/9cb0b3b4ac916cdf52d97a63d923dfbe73f0541e
> Cr-Commit-Position: refs/heads/master@{#14830}

TBR=sergeyu@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6513

Review-Url: https://codereview.webrtc.org/2464553002
Cr-Commit-Position: refs/heads/master@{#14831}
2016-10-29 00:14:16 +00:00
zijiehe
9cb0b3b4ac Add DesktopCapturer GetSourceList SelectSource FocusOnSelectedSource functions
I have chosen part of 2435603010 changes to compose this change.
According to the discussion we have made in previous change, this CL contains,
1. Source structure to represent a source of a DesktopCapturer.
2. GetSourceList / SelectSource / FocusOnSelectedSource functions in
DesktopCapturer.
3. ScreenCapturer and WindowCapturer forward corresponding functions to the new
DesktopCapturer APIs.

After this change, We can remove WindowCapturer & ScreenCapturer references from
Chromium, and use the new APIs.

BUG=webrtc:6513

Review-Url: https://codereview.webrtc.org/2452263003
Cr-Commit-Position: refs/heads/master@{#14830}
2016-10-29 00:07:00 +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
deadbeef
e7fc7d5c02 Fixing flaky DtmfSenderTest by using fake clock.
This test was expecting tones to be sent at specific times, with a 100ms
margin of error, causing slower bots or bots with less precise timing to
fail the test occasionally.

BUG=webrtc:4219,webrtc:5657

Review-Url: https://codereview.webrtc.org/2447013007
Cr-Commit-Position: refs/heads/master@{#14828}
2016-10-28 20:53:17 +00:00
ivoc
3e9a537601 Original CL: https://codereview.webrtc.org/2433153003/, commit 8b8d3e4c30e8ea3846b58dfd36d1fd35a7799df4.
Revert CL: https://codereview.webrtc.org/2456333002/, commit 48dfab5c58119a4e65c52506ed55f8de79725bcf.

The new function on the APM interface is no longer pure virtual.

BUG=webrtc:6525
TBR=solenberg@webrtc.org,peah@webrtc.org

Review-Url: https://codereview.webrtc.org/2458993002
Cr-Commit-Position: refs/heads/master@{#14827}
2016-10-28 14:55:39 +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
danilchap
e2a0177255 Style cleanups in rtp header extension traits:
renamed kName to kUri and make it more const.
remove IsSupportedBy to reduce header dependency.

BUG=webrtc:1994

Review-Url: https://codereview.webrtc.org/2457783005
Cr-Commit-Position: refs/heads/master@{#14825}
2016-10-28 14:09:04 +00:00
ivoc
af27ed01d7 Add algorithm for Residual Echo Detector.
This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values.

BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2419563003
Cr-Commit-Position: refs/heads/master@{#14824}
2016-10-28 14:04:08 +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
terelius
c4b9b9419e Revert of Replace FileWrapper with File (in audio_device) (patchset #3 id:40001 of https://codereview.webrtc.org/2386963003/ )
Reason for revert:
Speculative revert. This CL is a plausible cause for breakages in internal projects.

Original issue's description:
> Removes all uses of FileWrapper in audio_device.
>
> BUG=webrtc:6463
>
> Committed: https://crrev.com/04055e95bf97d106053d90bcc9e974eb4ad175e6
> Cr-Commit-Position: refs/heads/master@{#14811}

TBR=sprang@webrtc.org,henrika@webrtc.org,palmkvist@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6463

Review-Url: https://codereview.webrtc.org/2459873002
Cr-Commit-Position: refs/heads/master@{#14822}
2016-10-28 13:52:04 +00:00