Now does level estimate on the audio threads to avoid complex
copying of audio data to task queue. The old implementation could
also crash due to unclear ownership of the audio buffer.
BUG=webrtc:6569
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/2433393002 .
Cr-Commit-Position: refs/heads/master@{#14720}
The main goal of this CL is to remove old buffer handling using static arrays
and switch to the improved rtc::Buffer class instead.
By doing so, we can remove some members (since Buffer maintains them instead) and
do some additional cleanup.
This CL also fixes some minor style issues and improves the locking mechanism.
Finally, AudioDeviceBuffer::SetRecordingChannel() is deprecated since it has never been
used and is not included in any test.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2333273002
Cr-Commit-Position: refs/heads/master@{#14661}
Reason for revert:
There is a risk of ending up in a bad state due to race conditions with this patch. Tests in downstream clients have shown that it can
happen that an output stream is opened up in MUSIC mode when it should not.
Reverting since the new functionality added here is not worth the
risk of breaking existing clients.
Original issue's description:
> Android audio playout now supports non-call media streams.
>
> The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
>
> The solution is somewhat experimental.
>
> NOTRY=TRUE
>
> BUG=webrtc:4767
>
> Committed: https://crrev.com/872f614111f436d15e29516ce19c3b63d25b8639
> Cr-Commit-Position: refs/heads/master@{#14613}
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:4767
Review-Url: https://codereview.webrtc.org/2420583002
Cr-Commit-Position: refs/heads/master@{#14626}
The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
The solution is somewhat experimental.
NOTRY=TRUE
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2411263003
Cr-Commit-Position: refs/heads/master@{#14613}
Compromise solution where WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl() is marked
as deprecated and where as many APIs as possible that touches the HW AGC are removed. Some basic architecture is saved to ensure that we can restore usage of
the HW AGC if ever required for future devices.
The AppRTCMobile demo does still contain an AGC check box but it is now grayed out.
BUG=b/30387905
Review-Url: https://codereview.webrtc.org/2402883003
Cr-Commit-Position: refs/heads/master@{#14596}
RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.
This test ensures that audio starts pumping. The exact performance is
not that important.
R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True
Review-Url: https://codereview.webrtc.org/2391563002
Cr-Commit-Position: refs/heads/master@{#14492}
The former is always defined (by webrtc/base/checks.h) to either 0 or
1, whereas the latter isn't necessarily defined.
NOTRY=true
BUG=webrtc:6451
Review-Url: https://codereview.webrtc.org/2384693002
Cr-Commit-Position: refs/heads/master@{#14474}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
Trivial patch which fixes an issue where logged rate estimates could be
invalid. E.g. on iOS, two successive timer interrupts can be ~10.5 seconds
and not exactly 10.0 (which is usually the case on Android). In those
cases we could log a rate estimate of e.g. ~51000Hz instead of ~48000Hz.
This CL fixes that.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2350103002
Cr-Commit-Position: refs/heads/master@{#14305}
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.
Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).
More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.
BUG=webrtc:5925
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2119633004 .
Cr-Commit-Position: refs/heads/master@{#14290}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
This changes added a simple measurement of levels "close to the audio hardware"
both for playout and for recording. These levels are logged once each 10 seconds.
It also adds WebRTC.Audio.RecordedOnlyZeros UMA stat and it is updated at
destuction. It will report true iff all reported recording leves are zero.
BUG=NONE
R=peah@webrtc.org
Review URL: https://codereview.webrtc.org/2328433003 .
Cr-Commit-Position: refs/heads/master@{#14160}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Remove //build/config/sanitizers:deps as a dependency for
all rtc_executable targets and add it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2308553002
Cr-Commit-Position: refs/heads/master@{#14048}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
Reason for revert:
Seems to break an external client.
Original issue's description:
> Cleanup of the AudioDeviceBuffer class.
>
> WebRTC works on 10ms buffer sizes in both directions but this class has contained
> support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
>
> It also updates the style to follow the Google C++ style guide.
>
> Finally, I remove very old (not tested and not maintained) support for file
> handling since the code is never used. It was more or less dead code.
>
> BUG=NONE
> R=magjed@webrtc.org
>
> Committed: https://crrev.com/cf327b45b9f5738950d4fca2b6a7b6030d508cdf
> Cr-Commit-Position: refs/heads/master@{#13833}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=NONE
Review-Url: https://codereview.webrtc.org/2260183002
Cr-Commit-Position: refs/heads/master@{#13834}
WebRTC works on 10ms buffer sizes in both directions but this class has contained
support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly.
It also updates the style to follow the Google C++ style guide.
Finally, I remove very old (not tested and not maintained) support for file
handling since the code is never used. It was more or less dead code.
BUG=NONE
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2256833003 .
Cr-Commit-Position: refs/heads/master@{#13833}
When the sanitizer bots are switched to GN, this needs to be included as a dependency so that the executables can be compiled.
BUG=webrtc:6215
NOTRY=True
Review-Url: https://codereview.webrtc.org/2250893003
Cr-Commit-Position: refs/heads/master@{#13829}
This code does not work and hasn't been used in a long time. It also
lacks a GN target. There's no reason to save it.
BUG=none
Review-Url: https://codereview.webrtc.org/2255173002
Cr-Commit-Position: refs/heads/master@{#13820}
