Removes code duplication and use of the dangerous public destructor in
RefCountImpl.
Also making wider use of scoped_refptr and fixing various leaks in the
process.
BUG=webrtc:5229
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1477013005 .
Cr-Commit-Position: refs/heads/master@{#12075}
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}
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}
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}
- Change argument type to int for SetSendTelephoneEventPayloadType()
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1798903002
Cr-Commit-Position: refs/heads/master@{#11980}
- 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}
- 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}
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}
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}
- 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}
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}
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}
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}
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}
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: f14c47a58cTBR=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}
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}
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}
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}
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}
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}
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}
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}
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.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
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}
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}
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}
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}
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}
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}
