Commit Graph

631 Commits

Author SHA1 Message Date
94a23f04af Reland "Add check_deps rules in DEPS files."
Relanding https://codereview.webrtc.org/1796413002/
without the change to the openmax_dl include path
(which broke downstream code).

TBR=tommi@webrtc.org
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

Review URL: https://codereview.webrtc.org/1804333002 .

Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 11:05:50 +00:00
b69395b374 Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (patchset #2 id:20001 of https://codereview.webrtc.org/1802993002/ )
Reason for revert:
Revert because it breaks downstream code.

Original issue's description:
> Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/69a81999ace08e40e2b2ec526b0e111aa11b9538
> Cr-Commit-Position: refs/heads/master@{#12015}

TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1812453002

Cr-Commit-Position: refs/heads/master@{#12016}
2016-03-16 14:05:21 +00:00
69a81999ac Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1802993002

Cr-Commit-Position: refs/heads/master@{#12015}
2016-03-16 12:59:04 +00:00
56cf60e717 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
Reason for revert:
The openmax_dl include change breaks downstream projects.

Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}

TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623

Review URL: https://codereview.webrtc.org/1808573002

Cr-Commit-Position: refs/heads/master@{#12009}
2016-03-16 00:41:04 +00:00
086f851b7b Add check_deps rules in DEPS files.
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.

Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'

will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.

BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc

R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1796413002 .

Cr-Commit-Position: refs/heads/master@{#12008}
2016-03-16 00:22:53 +00:00
776593b139 Reland: Drop the 16kHz sample rate restriction on AECM and zero out higher bands
Landed originally here: https://codereview.webrtc.org/1774553002/
Revertede here: https://codereview.webrtc.org/1781893002/

TBR=solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1777093004

Cr-Commit-Position: refs/heads/master@{#12005}
2016-03-15 21:05:05 +00:00
6021fe2b1e Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1803923003

Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
e50872be13 Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1796183002

Cr-Commit-Position: refs/heads/master@{#11990}
2016-03-14 22:32:53 +00:00
1122dc0d9b Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Remove unused callback OnPlayTelephoneEvent from voe::Channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1804523002

Cr-Commit-Position: refs/heads/master@{#11984}
2016-03-14 18:52:33 +00:00
31642aa8f9 Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Change argument type to int for SetSendTelephoneEventPayloadType()

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1798903002

Cr-Commit-Position: refs/heads/master@{#11980}
2016-03-14 15:00:40 +00:00
b2a24ecf44 Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Clean up unused methods in voe::Channel following removal of VoEDtmf APIs.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1785643006

Cr-Commit-Position: refs/heads/master@{#11976}
2016-03-14 10:25:17 +00:00
b25345ee3f Replace scoped_ptr with unique_ptr in webrtc/call/
BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1789903003

Cr-Commit-Position: refs/heads/master@{#11970}
2016-03-12 14:10:53 +00:00
8842c3e41b Relanding https://codereview.webrtc.org/1715883002/ in pieces.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1782053002

Cr-Commit-Position: refs/heads/master@{#11953}
2016-03-11 11:06:48 +00:00
dfc2870380 Revert of Drop the 16kHz sample rate restriction on AECM and zero out higher bands (patchset #3 id:40001 of https://codereview.webrtc.org/1774553002/ )
Reason for revert:
Breaks Android it looks like.
See your own try jobs and
https://build.chromium.org/p/client.webrtc/builders/Android32%20Tests%20%28L%...

Original issue's description:
> Drop the 16kHz sample rate restriction on AECM and zero out higher bands
>
> The restriction has been removed completely and AECM now supports any
> number of higher bands. But this has been achieved by always zeroing out the
> higher bands, instead of applying a constant gain which is the average over half
> of the lower band (like it is done for the AEC), because that would be
> non-trivial to implement and we don't want to spend too much time on AECM, since
> we want to get rid of it in the long term anyway.
>
> R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org
>
> Committed: https://crrev.com/f687d53aabee0523ce6e9e0636163af8df120e41
> Cr-Commit-Position: refs/heads/master@{#11931}

TBR=peah@webrtc.org,turaj@webrtc.org,tina.legrand@webrtc.org,solenberg@webrtc.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1781893002

Cr-Commit-Position: refs/heads/master@{#11932}
2016-03-10 00:23:32 +00:00
f687d53aab Drop the 16kHz sample rate restriction on AECM and zero out higher bands
The restriction has been removed completely and AECM now supports any
number of higher bands. But this has been achieved by always zeroing out the
higher bands, instead of applying a constant gain which is the average over half
of the lower band (like it is done for the AEC), because that would be
non-trivial to implement and we don't want to spend too much time on AECM, since
we want to get rid of it in the long term anyway.

R=peah@webrtc.org, solenberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1774553002 .

Cr-Commit-Position: refs/heads/master@{#11931}
2016-03-09 15:38:09 +00:00
3ecb5c8698 Revert of - Clean up unused voice engine DTMF code. (patchset #4 id:60001 of https://codereview.webrtc.org/1722253002/ )
Reason for revert:
Breaks Chromium FYI bots for Android. E.g. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28K%20Nexus5%29/builds/4486/steps/content_browsertests/logs/stdio

Original issue's description:
> - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
> - Use better types in AudioSendStream::SendTelephoneEvent() and related methods.
>
> BUG=webrtc:4690
>
> Committed: https://crrev.com/8886c816582a7c6190c5429222cb8096fca302a6
> Cr-Commit-Position: refs/heads/master@{#11927}

TBR=tina.legrand@webrtc.org,henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1776243003

Cr-Commit-Position: refs/heads/master@{#11930}
2016-03-09 15:32:05 +00:00
8886c81658 - Clean up unused voice engine DTMF code following removal of VoEDtmf APIs.
- Use better types in AudioSendStream::SendTelephoneEvent() and related methods.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1722253002

Cr-Commit-Position: refs/heads/master@{#11927}
2016-03-09 11:32:53 +00:00
622d8950f5 Remove the VoEDtmf interface.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1723153002

Cr-Commit-Position: refs/heads/master@{#11906}
2016-03-08 12:11:00 +00:00
0e73934694 Remove webrtc/test/webrtc_test_common.gyp
Move the "webrtc_test_common" target to test.gyp and rename
it to "test_common".

Move all tests in "webrtc_test_common_unittests" (which
wasn't run on the bots) into "test_support_unittests".

NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1754593002

Cr-Commit-Position: refs/heads/master@{#11848}
2016-03-02 18:46:25 +00:00
7ffeab525c Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
This is a reland of https://codereview.webrtc.org/1737593002/ minus
the added missing headers in webrtc/{BUILD.gn,webrtc.gyp} and
webrtc/common.gyp that breaks GN in Chromium since it's using
the --check flag (which we should support).

BUG=webrtc:4243, webrtc:5589
TESTED=Tried generating GN files with --check in a Chromium checkout with this patch applied, successfully.
TBR=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1740873003 .

Cr-Commit-Position: refs/heads/master@{#11794}
2016-02-26 21:46:22 +00:00
7324eb9e62 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
Reason for revert:
Breaks GN in chromium.

Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}

TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589

Review URL: https://codereview.webrtc.org/1739783002

Cr-Commit-Position: refs/heads/master@{#11769}
2016-02-25 16:37:02 +00:00
3dd5d1d84a Remove PacketRouter sender distinction.
Instead relies on SetSendingMediaStatus() to filter out receiving RTP
modules. This status is now set in VoiceEngine's SetSend() for senders
along with SetSendingStatus().

BUG=
R=solenberg@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1705763002 .

Cr-Commit-Position: refs/heads/master@{#11768}
2016-02-25 15:56:58 +00:00
99b345c4e5 Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
depending on voice engine, resulting in a cyclic dependency (which we
don't detect since we have that check turned off, see webrtc:4243).

BUG=webrtc:4243, webrtc:5589
R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1737593002 .

Cr-Commit-Position: refs/heads/master@{#11766}
2016-02-25 14:12:48 +00:00
a26ac925f7 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.

Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}

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

Review URL: https://codereview.webrtc.org/1737013002

Cr-Commit-Position: refs/heads/master@{#11762}
2016-02-25 12:50:09 +00:00
da33a8a2a2 Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
Reason for revert:
Breaks Chromium.

Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58c

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

Review URL: https://codereview.webrtc.org/1736663004

Cr-Commit-Position: refs/heads/master@{#11761}
2016-02-25 12:34:12 +00:00
f14c47a58c Remove ignored return code from modules.
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1703833002 .

Cr-Commit-Position: refs/heads/master@{#11747}
2016-02-24 15:51:23 +00:00
b7f89d6e66 Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1702983002

Cr-Commit-Position: refs/heads/master@{#11658}
2016-02-17 18:04:26 +00:00
59013bcafb Remove spammy GetRTPStatistics() log.
BUG=webrtc:5442
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1695613003 .

Cr-Commit-Position: refs/heads/master@{#11596}
2016-02-12 10:35:18 +00:00
d8de1154c9 Remove mutable from rtc::CriticalSections.
A couple of mutables were added after last removal of mutables, so
removing those. rtc::CriticalSection is const-lockable.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1652983002

Cr-Commit-Position: refs/heads/master@{#11447}
2016-02-01 17:00:59 +00:00
bba9dec4d5 Use separate rtp module lists for send and receive in PacketRouter.
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.

Also moves sending transport feedback to the pacer thread.

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1628683002

Cr-Commit-Position: refs/heads/master@{#11443}
2016-02-01 12:40:04 +00:00
55b97fe388 clang-format -i -style=file webrtc/voice_engine/channel.*
This CL changes literally nothing else.

Review URL: https://codereview.webrtc.org/1644633005

Cr-Commit-Position: refs/heads/master@{#11416}
2016-01-28 13:22:52 +00:00
31fc21f454 Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
Also remove mischievous tab character!
This is a part of getting rid of CriticalSectionWrapper and makes the code slightly simpler.

BUG=

Review URL: https://codereview.webrtc.org/1607353002

Cr-Commit-Position: refs/heads/master@{#11346}
2016-01-21 18:37:44 +00:00
3313ec901f Enable transport seq num extension on receive channel to suppress log warning.
TBR=pbos@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1608563005

Cr-Commit-Position: refs/heads/master@{#11338}
2016-01-21 14:32:48 +00:00
429c345b02 Fixes a bug which incorrectly logs incoming RTCP as outgoing.
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1571283002

Cr-Commit-Position: refs/heads/master@{#11336}
2016-01-21 13:42:10 +00:00
d66b44d565 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741
Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
Cr-Commit-Position: refs/heads/master@{#11093}

Review URL: https://codereview.webrtc.org/1540103002

Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 11:06:41 +00:00
2d110be77f Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.

Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}

TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1588693002

Cr-Commit-Position: refs/heads/master@{#11241}
2016-01-13 20:00:29 +00:00
e591f9377f Storing raw audio sink for default audio track.
BUG=webrtc:5250

Review URL: https://codereview.webrtc.org/1551813002

Cr-Commit-Position: refs/heads/master@{#11230}
2016-01-13 00:45:33 +00:00
6955870806 Convert channel counts to size_t.
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
2016-01-13 00:26:55 +00:00
3842c5c7f7 Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.

BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1535963002 .

Cr-Commit-Position: refs/heads/master@{#11220}
2016-01-12 12:55:11 +00:00
e2976c87f7 Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.

This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.

The change also removes gtest_disable.h as an unused include from many
other files.

BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1547343002 .

Cr-Commit-Position: refs/heads/master@{#11150}
2016-01-04 21:44:16 +00:00
a4df27b671 Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ )
Reason for revert:
Compile error on Android needs to be fixed before relanding.

Original issue's description:
> Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
>
> The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
> Original review: https://codereview.webrtc.org/1413483003/
>
> The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.
>
> NOTRY=true
> TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
> BUG=webrtc:4741
>
> Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a
> Cr-Commit-Position: refs/heads/master@{#11093}

TBR=glaznev@webrtc.org,henrik.lundin@webrtc.org,solenberg@google.com,henrikg@webrtc.org,perkj@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1537213002

Cr-Commit-Position: refs/heads/master@{#11094}
2015-12-19 18:14:18 +00:00
f4f5cb0927 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4.
Original review: https://codereview.webrtc.org/1413483003/

The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function.

NOTRY=true
TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1541633002

Cr-Commit-Position: refs/heads/master@{#11093}
2015-12-19 18:02:39 +00:00
36d4c54500 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ )
Reason for revert:
Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome.

Original issue's description:
> Added option to specify a maximum file size when recording an AEC dump.
>
> For applications with a strict filesize limit for debug files,
> I added an option to specify a maximum filesize for AEC dumps. An
> existing unit test is extended to check that the feature works as
> advertised.
>
> BUG=webrtc:4741
> TBR=glaznev@webrtc.org
>
> Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87
> Cr-Commit-Position: refs/heads/master@{#11081}

TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1533913004

Cr-Commit-Position: refs/heads/master@{#11087}
2015-12-18 16:05:21 +00:00
ae2c5ad12a Added option to specify a maximum file size when recording an AEC dump.
For applications with a strict filesize limit for debug files,
I added an option to specify a maximum filesize for AEC dumps. An
existing unit test is extended to check that the feature works as
advertised.

BUG=webrtc:4741
TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1413483003

Cr-Commit-Position: refs/heads/master@{#11081}
2015-12-18 11:53:42 +00:00
44307630d3 AudioCodingModuleImpl: Stop failing artificially for non-Opus encoders
All encoders already handle the "Opus-specific" requests sanely (by
failing nicely), so we don't need extra checks to protect them.

BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1527453005

Cr-Commit-Position: refs/heads/master@{#11051}
2015-12-16 14:24:09 +00:00
66085beef8 Bugfix that fixes the error where the audio processing module is called
using the wrong sample rate for the render signal.

The CL is basically a partial revert of the related changes done on
output_mixer.cc in the CL https://codereview.webrtc.org/1234463003.

The CL also reverts the removal of the input_sample_rate_hz() method
that was removed as part of the CL
https://codereview.webrtc.org/1379123002 (as it was at that point
no longer used).

It should be noted that this CL turns off the effect of the
IntelligibilityEnhancer when the AudioFrame AudioProcessing APIs are
used. While it may be possible to solve that by adding upsampling after
the API call, that  approach was discarded due to that:
-That would add extra processing in the echo path, leading to possible
AEC performance reduction.
-That would add extra complexity for the mobile case.
-That would only patch the intelligibility enhancer operation as the
proper way to do such an operation is within APM.
-The intelligibility enhancer is not active by default anywhere.

BUG=webrtc:5237

Review URL: https://codereview.webrtc.org/1525173002

Cr-Commit-Position: refs/heads/master@{#11045}
2015-12-16 10:02:26 +00:00
e5ae6f8237 Correcting the check for the return code produced by
AudioProcessing::ProcessReverseStream().
Before the change, only -1 was considered to be an error.

Allthough the error code scheme for AudioProcessing definitely
could be discussed, the current scheme have many error codes that
differ from -1 and thus were not caught by the old code.

BUG=webrtc:5237

Review URL: https://codereview.webrtc.org/1515073004

Cr-Commit-Position: refs/heads/master@{#11003}
2015-12-14 07:02:40 +00:00
f888bb58da Support for unmixed remote audio into tracks.
BUG=chromium:121673
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1505253004 .

Cr-Commit-Position: refs/heads/master@{#10995}
2015-12-12 00:37:14 +00:00
5c1def8892 modules/rtp_rtcp/include folder cleared of lint warnings
Functions that do not follow lint are marked deprecated, including function in the interface.

BUG=webrtc:5308
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1493403003

Cr-Commit-Position: refs/heads/master@{#10975}
2015-12-10 17:52:01 +00:00
84e78f9102 Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)

Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.

Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.

BUG=webrtc:5177

Review URL: https://codereview.webrtc.org/1457023002

Cr-Commit-Position: refs/heads/master@{#10965}
2015-12-10 09:51:02 +00:00