Before this change, the APM limiter used in FrameCombiner (a
sub-component of AudioMixer) only gets to process the data when the
number of non-muted streams is >1. If this number varies between <=1
and >1, the limiter's view of the data will have gaps during the
periods with <= 1 active stream.
This leads to discontinuities in the applied gain. These
discontinuities cause clicks in the output audio. This change
activates APM limiter processing based on the number of audio streams,
independently of their mutedness status.
BUG=chromium:695993
Review-Url: https://codereview.webrtc.org/2776113002
Cr-Commit-Position: refs/heads/master@{#17442}
The APM limiter is a component for keeping the audio from clipping by smoothly reducing the amplitude of the audio samples. It can be rather expensive because of band-splitting & merging. Also, experiments indicate that it is of questionable benefit (adding several sources of human speech almost never cause clipping).
To optionally disable the limiter, this CL does some refactoring on the (quite large) AudioMixerImpl. Functionality related to actual addition of frames and handling AudioFrame meta-data (sample_rate, num_channels, samples_per_channel, time_stamp, elapsed_time_ms) is broken out in a new sub-component called FrameCombiner.
The FrameCombiner is initialized with a 'use_limiter' flag. To create a mixer without using the APM limiter
Inside of FrameCombiner, the meta-data handling and the audio sample addition are kept divided from each other.
This also fixes a few minor GN issues so that warnings do not have to be suppressed.
BUG=webrtc:7167
Review-Url: https://codereview.webrtc.org/2692333002
Cr-Commit-Position: refs/heads/master@{#16742}
This CL breaks out the output sample rate calculation from
webrtc::AudioMixerImpl. A new OutputRateCalculator interface is added
to make the sample rate configurable. There are at least three reasons
for this change:
1. The mixer will be used for an internal project, in which no
resampling is done after the mixing. There the sample rate should
be static. Currently, it can differ across mix iterations and
depends on the number of audio sources. If there are no sources,
the WebRTC mixer behavior is to produce silence at 48 kHz.
2. A planned change to WebRTC will make audio processing steps
happen at constant sample rates. A configurable sample rate
calculator will make the transition simpler for the mixer.
3. The current mixer design is a single large file. Behavior is not
always simple to change (e.g. as in this case to mix at a
constant rate), unrelated behavior can be broken, reusing the
mixer in internal projects is tricky. Using DI for the sample
rate calculation solves parts of these issues.
Changes:
The protected mixer c-tor now takes
unique_ptr<OutputRateCalculator>. The current output rate calculation
is moved to DefaultOutputRateCalculator. A new factory method
AudioMixerImpl::CreateWithOutputRateCalculator is added. The old
factory method passes the default rate calculator.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2557713006
Cr-Commit-Position: refs/heads/master@{#15472}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
This is a refactoring change in preparation for enabling AudioMixer
with the goal to have a small CL as possible for passing audio through
the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/
enables the mixer.
An object of class AudioState is shared across different webrtc audio
connections. It is created in tests and in
WebRTCVoiceEngine. AudioState is constructed by passing a Config
struct, where one argument is scoped_refptr<AudioMixer>.
Populating this field has now been mandatory. Tests and
WebRTCVoiceEngine create and pass either a AudioMixerImpl.
WebRTCVoiceEngine passes a real AudioMixer, which is
currently unused.
An alternative would have tests pass a mocked audio mixer. We
chose not to do that, because we believe that tests should use
the real thing unless there are reasons against it. Construction
time is not an issue, because the real mixer is relatively
lightweight.
We couldn't find a way to test any mixer-related changes in AudioState
before the mixes is connected. The next dependent CL
https://codereview.webrtc.org/2436033002/ contains unit tests for
mixer usage.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2469743002
Cr-Commit-Position: refs/heads/master@{#15134}
Introduce a convention on categorizing GN targets:
1. Production code
2. Tests
3. Examples
4. Tools
The first two have targets spread out all over the tree,
while the latter are isolated to examples/ and tools/ directories.
Another new convention: Each directory's BUILD.gn file shall contain
a target named similar to the directory name. This target shall
contain the 'most common' production code, i.e. so that most
consumers of the directory can depend on only the directory
(which implicitly means that target in GN).
//webrtc:webrtc_tests is changed to depend on all WebRTC tests.
From now on, it's necessary to add new test targets to this dependency
tree in order to get them compiled.
Two new group targets are created:
//webrtc/modules/audio_coding:audio_coding_tests
//webrtc/modules/audio_processing:audio_processing_tests
to reduce the long list of tests in //webrtc:webrtc_tests.
Visibility on //webrtc:webrtc and //webrtc:webrtc_tests is restricted
to the root target, to avoid circular dependencies due to the monolithic
property of these targets (a problem we've had in the past).
The 'root' target at the top level is renamed to 'default', which means GN will
build this target instead of _all_ generated targets
(see https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/faq.md#Can-I-control-what-targets-are-built-by-default).
This target now depends on everything we want to build, meaning all targets now
explicitly needs to be wired up from the root target in order to get build.
Having this, the number of compiled objects on Android is decreased from 8855 to 6276. It also gives us better control over our build.
BUG=webrtc:6440
TESTED=git cl try --clobber
NOTRY=True
Review-Url: https://codereview.webrtc.org/2441383002
Cr-Commit-Position: refs/heads/master@{#14821}
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
MixerAudioSource is moved to AudioMixerImpl::Source. Structures and methods of the MixerAudioSource interface have been renamed. The RemixFrame method has added checks and is moved to audio_frame_manipulator.h
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396803004
Cr-Commit-Position: refs/heads/master@{#14600}
Reason for revert:
breaks chromium FYI
Original issue's description:
> Made MixerAudioSource a pure interface.
>
> This required quite a few small changes in the mixing algorithm
> structure, the mixer interface and the mixer unit tests.
>
> BUG=webrtc:6346
>
> Committed: https://crrev.com/2ae5fdff86b784545cbd724de54bb5ffedde1adf
> Cr-Commit-Position: refs/heads/master@{#14567}
TBR=ivoc@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2394253003
Cr-Commit-Position: refs/heads/master@{#14568}
This required quite a few small changes in the mixing algorithm
structure, the mixer interface and the mixer unit tests.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396483002
Cr-Commit-Position: refs/heads/master@{#14567}
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}
1. Use of const in all variable declarations where it is possible
2. Variable names and function arguments changed from CamelCase to match code style
3. A few stale comments removed.
4. Chromium clang plugin check added (now possible thanks to kwiberg@'s work on common.h)
5. Disallow constructor macros added.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2294263002
Cr-Commit-Position: refs/heads/master@{#14120}
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}
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}
Removed the OutputMixer part of the new mixer and renamed the new
mixer from NewAudioConferenceMixer to AudioMixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2249213005
Cr-Commit-Position: refs/heads/master@{#13883}
Added a level indicator to the new mixer. The level indicator is
webrtc::voe::AudioLevel. It computes the current audio level, which is
used all the way up to peerconnection.
This is part of the project to rewrite the old conference mixer and
output mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2230823004
Cr-Commit-Position: refs/heads/master@{#13878}
Also fixed one small chromium-style error in the new mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2234293002
Cr-Commit-Position: refs/heads/master@{#13752}
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort. Also added some dependencies that were previously in
another part of the tree.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
Removed resampler from NewAudioConferenceMixer and AudioMixer (which started as a copy of former OutputMixer). This is part of the mixer rewrite project. In particular,
this is one of the steps required to have a single mixing component
instead of two doing the same thing.
The next planned change (which is not part of this CL) is to plug in the
new mixer (NewAudioConferenceMixer) into AudioState and AudioDeviceModule.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2221443002
Cr-Commit-Position: refs/heads/master@{#13674}
I have added build files and renamed the mixer so that it doesn't conflict with the old one. The header includes now point to this copy of the mixer. I have also fixed some of the more obvious cases of style guide non-conformance and run 'PRESUBMIT' on the old mixer.
This is a first step in the creation of a new mixing module that will replace AudioConferencMixer and OutputMixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2104363003
Cr-Commit-Position: refs/heads/master@{#13378}
