Commit Graph

5069 Commits

Author SHA1 Message Date
d153a37801 Remove contention between RTCP packets and encoding.
Receiving RTCP often caused the worker thread to stall for >20 ms
(>100ms observed) due to contention on VideoSender's send_crit_ (used to
protect encoding).

This change removes an unnecessary acquire of send_crit_ and caches
encoder settings in ViEEncoder instead of acquiring them through
::SendCodec() in VCM (which is blocking).

BUG=webrtc:5106
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1433703002 .

Cr-Commit-Position: refs/heads/master@{#10582}
2015-11-10 15:27:21 +00:00
cfc319be1d Reland of Work on flexible mode and screen sharing. (patchset #1 id:1 of https://codereview.webrtc.org/1438543002/ )
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..

(This time from my @webrtc account.)

Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1431283002

Cr-Commit-Position: refs/heads/master@{#10581}
2015-11-10 15:17:26 +00:00
c95c366f5a Move the Rent-A-Codec™ from CodecOwner to CodecManager
Future CLs will move it even further down the stack.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1431103002

Cr-Commit-Position: refs/heads/master@{#10580}
2015-11-10 14:35:28 +00:00
0be8f1d347 Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
Reason for revert:
Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.

Original issue's description:
> Work on flexible mode and screen sharing.
>
> Implement VP8 style screensharing but with spatial layers.
> Implement flexible mode.
>
> Files from other patches:
> generic_encoder.cc
> layer_filtering_transport.cc
>
> BUG=webrtc:4914
>
> Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> Cr-Commit-Position: refs/heads/master@{#10572}

TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1438543002

Cr-Commit-Position: refs/heads/master@{#10578}
2015-11-10 13:31:22 +00:00
Per
327d8babc8 Add DecodedImageCallback::Decoded() function with custom decode time value.
On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.

This new method will be used in
https://codereview.webrtc.org/1422963003/

BUG=webrtc:4993
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1414693006 .

Cr-Commit-Position: refs/heads/master@{#10576}
2015-11-10 13:00:45 +00:00
805fc710f7 Let Rent-A-Codec™ create and own speech encoders
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1410333015

Cr-Commit-Position: refs/heads/master@{#10575}
2015-11-10 12:05:23 +00:00
3cea256806 Reland "Prevent Opus DTX from generating intermittent noise during silence"
The original CL is reviewed at
https://codereview.webrtc.org/1415173005/

A silly mistake was made at the last patch set, and the CL was reverted. This CL is to fix and reland it.

BUG=

Review URL: https://codereview.webrtc.org/1422213003

Cr-Commit-Position: refs/heads/master@{#10574}
2015-11-10 11:49:32 +00:00
77ccfb4d16 Work on flexible mode and screen sharing.
Implement VP8 style screensharing but with spatial layers.
Implement flexible mode.

Files from other patches:
generic_encoder.cc
layer_filtering_transport.cc

BUG=webrtc:4914

Review URL: https://codereview.webrtc.org/1328113004

Cr-Commit-Position: refs/heads/master@{#10572}
2015-11-10 10:19:20 +00:00
ce83ae1c19 Improve informative message in codereview.settings.
In https://codereview.webrtc.org/1389963002 the message
displayed when trying to create a CL from an unsupported
location was improved. However it's confusing for developers
working from a WebRTC checkout if they stand in src/webrtc
when trying to create a CL.

R=henrika@webrtc.org, phoglund@webrtc.org

Review URL: https://codereview.webrtc.org/1432073002 .

Cr-Commit-Position: refs/heads/master@{#10571}
2015-11-10 10:08:36 +00:00
c12be3984f -Removed the indirect error message reporting in aec and aecm.
-Made the component error messages generic to be an unspecified error message.

BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1404743003

Cr-Commit-Position: refs/heads/master@{#10570}
2015-11-10 07:53:53 +00:00
952892a28a Fix a 64-bit pointer truncation bug found by VC++ 2015
When converting from void* to unsigned long long it is dangerous to go
through unsigned long because for VC++ 64-bit builds this will be 32
bits. When casting a pointer to an integral type the safest type to
choose for the integral cast is always intptr_t or uintptr_t.

BUG=440500
NOPRESUBMIT=true

Review URL: https://codereview.webrtc.org/1437433002

Cr-Commit-Position: refs/heads/master@{#10569}
2015-11-10 06:52:00 +00:00
b4a753fdb5 Revert of Prevent Opus DTX from generating intermittent noise during silence (patchset #10 id:250001 of https://codereview.webrtc.org/1415173005/ )
Reason for revert:
Breaks voe_auto_test on all three "large tests bots".
https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/5630/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Mac32%20Release%20%5Blarge%20tests%5D/builds/5599/steps/voe_auto_test/logs/stdio
https://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/5645/steps/voe_auto_test/logs/stdio

Notice these bots are no longer a part of the default trybot set, so they have to be run manually when working with code that affects their tests (they were removed as they queued up all the time in the CQ, and usually don't catch breakages).

Original issue's description:
> Prevent Opus DTX from generating intermittent noise during silence.
>
> Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
>
> BUG=webrtc:5127
>
> Committed: https://crrev.com/f475add57eada116bc960fe2935876ec8c077977
> Cr-Commit-Position: refs/heads/master@{#10565}

TBR=tina.legrand@webrtc.org,kwiberg@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1428613004

Cr-Commit-Position: refs/heads/master@{#10567}
2015-11-09 21:27:11 +00:00
c1cd2bbd79 Turned off progress report for finished processing when the progress report is explicitly deactivated
BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1407723002

Cr-Commit-Position: refs/heads/master@{#10566}
2015-11-09 18:38:12 +00:00
f475add57e Prevent Opus DTX from generating intermittent noise during silence.
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.

BUG=webrtc:5127

Review URL: https://codereview.webrtc.org/1415173005

Cr-Commit-Position: refs/heads/master@{#10565}
2015-11-09 18:08:20 +00:00
ab48ef3534 Remove legacy audio device glue files.
I cannot find any references to these old locations.

TESTED=
git cl try -c --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1421723009 .

Cr-Commit-Position: refs/heads/master@{#10564}
2015-11-09 16:44:05 +00:00
48407f7122 Changed queue implementation to the proposed vector-based solution.
Added unit tests.

BUG=webrtc:5099
TBR=hlundin-webrtc

Review URL: https://codereview.webrtc.org/1398473004

Cr-Commit-Position: refs/heads/master@{#10562}
2015-11-09 13:24:56 +00:00
1f1912d1f0 Added unittest of the locking functionality in the audio processing module
The test is currently disabled as it takes too long to run in a coffe-cup manner

BUG=webrtc:5099

Review URL: https://codereview.webrtc.org/1394803002

Cr-Commit-Position: refs/heads/master@{#10560}
2015-11-09 11:13:25 +00:00
39d8bee397 Make ACMCodecDB private to RentACodec
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1414203010

Cr-Commit-Position: refs/heads/master@{#10549}
2015-11-07 00:22:50 +00:00
566ef247b9 Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1403363003

Cr-Commit-Position: refs/heads/master@{#10548}
2015-11-06 23:34:58 +00:00
19299fb28b Remove interface directories kept to avoid breaking downstream.
This is a follow-up CL for https://codereview.webrtc.org/1417683006
now that downstream code has been updated to use the 'include' directories
for header files instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel

Review URL: https://codereview.webrtc.org/1414793020

Cr-Commit-Position: refs/heads/master@{#10547}
2015-11-06 23:24:52 +00:00
d6c0f8cac1 Remove ACMCodecDB::Codec, and make the rest of ACMCodecDB private
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1423043005

Cr-Commit-Position: refs/heads/master@{#10546}
2015-11-06 22:28:08 +00:00
23725e09c6 Remove ICU usage from jni_helpers.cc.
JNI already has jstring<->UTF8 string conversion, so using that should
save ~1mb off android binaries (ICU is *large*), probably around
300-400k after compression.

BUG=

Review URL: https://codereview.webrtc.org/1430023005

Cr-Commit-Position: refs/heads/master@{#10545}
2015-11-06 21:56:11 +00:00
d66daa2d2f Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
BUG=

Review URL: https://codereview.webrtc.org/1419523004

Cr-Commit-Position: refs/heads/master@{#10542}
2015-11-06 17:00:21 +00:00
56b1128c8f Change to use local Random object instead of global rand() in the RtcEventLog unit test.
Removed Rand(int low, int high) since that function outputs results that are non-random and/or outside the interval if low is negative.

Added new Uniform(uint32_t, uint32_t) function to replace Rand(int low, int high).

Changed various unit tests to use the new functions.
BUG=

Review URL: https://codereview.webrtc.org/1413053002

Cr-Commit-Position: refs/heads/master@{#10541}
2015-11-06 13:14:01 +00:00
c4a1c370aa Removed vie_defines.h
The defines still in use was only used in single files, so they were
moved to these specific cc-files.

Review URL: https://codereview.webrtc.org/1411573007

Cr-Commit-Position: refs/heads/master@{#10539}
2015-11-06 12:33:56 +00:00
dc0da59eba Remove old system_wrappers event_tracer.h.
Replaced by webrtc/base/event_tracer.h and no longer used.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1421463008

Cr-Commit-Position: refs/heads/master@{#10537}
2015-11-06 10:16:56 +00:00
fb3d8b3df2 Remove ACMCodecDB::CodecFreq
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1408773005

Cr-Commit-Position: refs/heads/master@{#10536}
2015-11-06 09:24:16 +00:00
288886b2ec Pass audio to AudioEncoder::Encode() in an ArrayView
Instead of in separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1418423010

Cr-Commit-Position: refs/heads/master@{#10535}
2015-11-06 09:21:39 +00:00
c253a1c00e Reland of "Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile."
BUG=webrtc:5144
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1409753007

Cr-Commit-Position: refs/heads/master@{#10533}
2015-11-06 08:12:09 +00:00
b7a5c16d2c Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
This is the second revert. The first attempt in https://codereview.webrtc.org/1423693008/
was missing a subtle curly brace caused by a merge conflict.
I'm going to let this one go through the CQ.

Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.

See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.

Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1412963007

Cr-Commit-Position: refs/heads/master@{#10532}
2015-11-05 20:33:25 +00:00
006d93d3c6 Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
BUG=

Review URL: https://codereview.webrtc.org/1411673003

Cr-Commit-Position: refs/heads/master@{#10531}
2015-11-05 20:02:19 +00:00
962c5ce7e8 Re-enable VP9 resize test.
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1409973009 .

Cr-Commit-Position: refs/heads/master@{#10530}
2015-11-05 18:11:49 +00:00
93a2febe73 Move ACMCodecDB::ValidPayloadType to RentACodec
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1430043003

Cr-Commit-Position: refs/heads/master@{#10525}
2015-11-05 15:39:42 +00:00
86b40506b3 Reland of Add aecdump support to audioproc_f. (patchset #2 id:250001 of https://codereview.webrtc.org/1423693008/ )
Reason for revert:
Oh dear, this broke compilation.
I guess more was built on top of this CL before I reverted it.

Reverting now for futher investigation (and re-land using CQ)

Original issue's description:
> Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
>
> Reason for revert:
> This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
> I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.
>
> See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.
>
> Original issue's description:
> > Add aecdump support to audioproc_f.
> >
> > Add a new interface to abstract away file operations. This CL temporarily
> > removes support for dumping the output of reverse streams. It will be easy to
> > restore in the new framework, although we may decide to only allow it with
> > the aecdump format.
> >
> > We also now require the user to specify the output format, rather than
> > defaulting to the input format.
> >
> > TEST=Bit-exact output to the previous audioproc_f version using an input wav
> > file, and to the legacy audioproc using an aecdump file.
> >
> > Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> > Cr-Commit-Position: refs/heads/master@{#10460}
>
> TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/d279941bb54bfdc6e7324bf36cac76581474b96d
> Cr-Commit-Position: refs/heads/master@{#10523}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1419953010

Cr-Commit-Position: refs/heads/master@{#10524}
2015-11-05 14:23:10 +00:00
d279941bb5 Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ )
Reason for revert:
This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios
I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions.

See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve.

Original issue's description:
> Add aecdump support to audioproc_f.
>
> Add a new interface to abstract away file operations. This CL temporarily
> removes support for dumping the output of reverse streams. It will be easy to
> restore in the new framework, although we may decide to only allow it with
> the aecdump format.
>
> We also now require the user to specify the output format, rather than
> defaulting to the input format.
>
> TEST=Bit-exact output to the previous audioproc_f version using an input wav
> file, and to the legacy audioproc using an aecdump file.
>
> Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08
> Cr-Commit-Position: refs/heads/master@{#10460}

TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1423693008

Cr-Commit-Position: refs/heads/master@{#10523}
2015-11-05 14:09:08 +00:00
394c537b21 Update layer indices for non-flexible mode according to updates in the RTP payload profile.
https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1426813002

Cr-Commit-Position: refs/heads/master@{#10522}
2015-11-05 14:07:09 +00:00
f97bfed6c7 Revert of Move audioproc_test_utils into enable_protobuf condition. (patchset #1 id:1 of https://codereview.webrtc.org/1419533010/ )
Reason for revert:
I'm going to revert this and https://codereview.webrtc.org/1409943002 to get things back in a good state.

Original issue's description:
> Move audioproc_test_utils into enable_protobuf condition.
>
> From https://codereview.webrtc.org/1409943002 this target doesn't
> pass GYP on iOS unless build_with_libjingle==1.
> Our bots currently build with that, but we want to remove that GYP_DEFINES
> from the bots since http://www.webrtc.org/native-code/ios doesn't
> say it's needed.
>
> R=aluebs@webrtc.org
>
> Committed: https://crrev.com/e2a89251d9a75c2439daddd80f732ab505d0e1b9
> Cr-Commit-Position: refs/heads/master@{#10510}

TBR=andrew@webrtc.org,aluebs@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1428063007

Cr-Commit-Position: refs/heads/master@{#10521}
2015-11-05 14:02:20 +00:00
cd19faffa8 Attempt to isolate a bug by adding a new CHECK
Review URL: https://codereview.webrtc.org/1426953005

Cr-Commit-Position: refs/heads/master@{#10520}
2015-11-05 13:11:26 +00:00
2f48d94124 Set pacer target bitrate to max of bwe and bitrate allocation.
ChannelGroup::OnNetWorkChanged() should not configure the pacer to send
a lower bitrate than what bitrate_allocator has actually allocated (may
be the case if min_bitrate is enforced, for instance).

BUG=

Review URL: https://codereview.webrtc.org/1413663004

Cr-Commit-Position: refs/heads/master@{#10519}
2015-11-05 12:25:58 +00:00
d6b9d3353d Moves logging of audio effects to when they are enabled
BUG=none
TBR=magjed

Review URL: https://codereview.webrtc.org/1411783011 .

Cr-Commit-Position: refs/heads/master@{#10516}
2015-11-05 11:44:40 +00:00
444237e2e4 Remove unused crypto.gni import.
We are trying to move crypto.gni out of build/config in chromium,
this should help with this work.

BUG=None
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1410343014

Cr-Commit-Position: refs/heads/master@{#10514}
2015-11-04 22:44:04 +00:00
cecd7b8301 Disable VP9 resize test.
Needed for the upcoming libvpx roll.
Will re-enable it after the roll.

TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1421603004 .

Cr-Commit-Position: refs/heads/master@{#10513}
2015-11-04 22:35:56 +00:00
9b66957b4f Stop a session when a new connection becomes writable.
We cannot do it at the end of sorting because it may stop a session too early.
Also remove was_writable_, which is not useful.
BUG=webrtc:5119

Review URL: https://codereview.webrtc.org/1406423008

Cr-Commit-Position: refs/heads/master@{#10511}
2015-11-04 20:07:49 +00:00
e2a89251d9 Move audioproc_test_utils into enable_protobuf condition.
From https://codereview.webrtc.org/1409943002 this target doesn't
pass GYP on iOS unless build_with_libjingle==1.
Our bots currently build with that, but we want to remove that GYP_DEFINES
from the bots since http://www.webrtc.org/native-code/ios doesn't
say it's needed.

R=aluebs@webrtc.org

Review URL: https://codereview.webrtc.org/1419533010 .

Cr-Commit-Position: refs/heads/master@{#10510}
2015-11-04 18:56:58 +00:00
98cc88c873 Correctly handle the error case where the CodecId has a negative value
Negative values should be treated the same as too-large positive values: by returning an empty Maybe.

TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1407383010

Cr-Commit-Position: refs/heads/master@{#10509}
2015-11-04 17:56:22 +00:00
5d4e944391 Revert of Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. (patchset #3 id:40001 of https://codereview.webrtc.org/1427253002/ )
Reason for revert:
Breaks bot.

Original issue's description:
> Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
>
> Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.
>
> Size of WebRtcRTPHeader: 4352 -> 1784 bytes.
>
> BUG=webrtc:5144, chromium:500602
>
> Committed: https://crrev.com/81c5c7f8157f767747bd97419eb0a589207354cf
> Cr-Commit-Position: refs/heads/master@{#10504}

TBR=stefan@webrtc.org,mflodman@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5144, chromium:500602

Review URL: https://codereview.webrtc.org/1423493005

Cr-Commit-Position: refs/heads/master@{#10508}
2015-11-04 15:40:44 +00:00
e56c763691 Removing trace checks in VoETestManager.
Trace checks in the ctor and dtor of VoETestManager are removed, since they can fail if there are more than one VoETestManager (or VoE) used in a test.

BUG=

Review URL: https://codereview.webrtc.org/1407883007

Cr-Commit-Position: refs/heads/master@{#10507}
2015-11-04 15:11:52 +00:00
275d255e21 Adding debug dump test.
This test is to verify that the debug dump can perfectly reproduce APM states if the recording is made from the first input sample.

BUG=

Review URL: https://codereview.webrtc.org/1393353003

Cr-Commit-Position: refs/heads/master@{#10506}
2015-11-04 14:24:02 +00:00
81c5c7f815 Change type of pid_diff (int16_t -> uint8_t) according to updates in RTP payload profile. Max p_diff is 8 bits.
Change type of number of reference pictures (size_t -> uint8_t). Max is 2 bits.

Size of WebRtcRTPHeader: 4352 -> 1784 bytes.

BUG=webrtc:5144, chromium:500602

Review URL: https://codereview.webrtc.org/1427253002

Cr-Commit-Position: refs/heads/master@{#10504}
2015-11-04 10:19:45 +00:00
f040b2367d Add histograms for send-side delay stats for a sent video stream:
- "WebRTC.Video.SendSideDelayInMs"
- "WebRTC.Video.SendSideDelayMaxInMs"

BUG=chromium:512752

Review URL: https://codereview.webrtc.org/1405023014

Cr-Commit-Position: refs/heads/master@{#10502}
2015-11-04 08:59:10 +00:00