Along the way slightly simplify the class interface since views
carry audio properties. Also, now allocating FrameCombiner allocates
the mixing buffer in the same allocation.
Bug: chromium:335805780
Change-Id: Id7a76b040c11064e1e4daf01a371328769162554
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352502
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42465}
Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
New ctor added without optional and media specific fields.
Bug: webrtc:10739, b/246753278
Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38124}
Users of the mixer can use this information to determine which sources were included in the frame.
Bug: webrtc:12745
Change-Id: I11a8e3b1f4e8f95eb870336cad8dd082330bdf02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217768
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34035}
Made path from NetEq to AudioTransport ready for many-channel audio.
If there is one stream, we can handle anything that fits in an
AudioFrame. For many streams, the current limit is 6.
Some multi-channel combinations are not supported: e.g. if we get
stereo audio and attempt to play out 6 channels.
Changes:
* AudioFrameOperations - replaced the MonoTo* and *ToMono methods by
UpmixChannels & DownmixChannels.
* AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing
algorithm to handle many channels.
Bug: webrtc:8649
Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb
Reviewed-on: https://webrtc-review.googlesource.com/c/106040
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26446}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
The FrameCombiner sub-module of the AudioMixer uses one of two
limiters. One is an AudioProcessingModule with AGC1 enabled and
configured as a limiter. The other is the limiter part of AGC2. This
change removes the APM-AGC1 limiter. This requires small changes to
FrameCombiner, AudioMixerImpl and tests.
We also stop using the finch experiment flag.
Bug: webrtc:8925
Change-Id: Id7b8349ec4720b6417b15eaf70ed1a850b6ddbed
Reviewed-on: https://webrtc-review.googlesource.com/84620
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23727}
The webrtc::AudioMixer uses a limiter component. This CL allows
changes the APM-AGC limiter to the APM-AGC2 limiter though a Chrome
field trial.
The AGC2 limiter has a float interface. We plan to eventually switch
to the AGC2 limiter. Therefore, we will now mix in de-interleaved
floats. Float mixing will happen both when using the old limiter and
when using the new one.
After this CL the mixer will support two limiters. The limiters have
different interfaces and need different processing steps. Because of
that, we make (rather big) changes to the control flow in
FrameCombiner. For a short while, we will mix in deinterleaved floats
when using any limiter.
Originally landed in https://webrtc-review.googlesource.com/c/src/+/56141/
Reverted in https://webrtc-review.googlesource.com/c/src/+/57940
because of both breaking compilation and having a severe error. The
error is fixed and a test is added. The compilation issue is fixed.
Bug: webrtc:8925
Change-Id: Ieba138dee9652c826459fe637ae2dccbbc06bcf0
Reviewed-on: https://webrtc-review.googlesource.com/58085
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22207}
This reverts commit bd7b461f16.
Reason for revert: Broke the internal project. The issue maybe related to the apm_debug_dump configuration.
Original change's description:
> Choose between APM-AGC-Limiter and Apm-AGC2-fixed-gain_controller.
>
> The webrtc::AudioMixer uses a limiter component. This CL changes the
> APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
>
> The new limiter has a float interface. Since we're moving to it, we
> now mix in floats as well. After this CL the mixer will support two
> limiters. The limiters have different interfaces and need different
> processing steps. Because of that, we make (rather big) changes to the
> control flow in FrameCombiner. For a short while, we will mix in
> deinterleaved floats when using any limiter.
>
> NOTRY=true
>
> Bug: webrtc:8925
> Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
> Reviewed-on: https://webrtc-review.googlesource.com/56141
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22185}
TBR=gustaf@webrtc.org,aleloi@webrtc.org
Change-Id: I3dd1a2b1fca32c4dd046e6fc325744079e3ac5ca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8925
Reviewed-on: https://webrtc-review.googlesource.com/57940
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22189}
The webrtc::AudioMixer uses a limiter component. This CL changes the
APM-AGC limiter to the APM-AGC2 limiter though a Chrome field trial.
The new limiter has a float interface. Since we're moving to it, we
now mix in floats as well. After this CL the mixer will support two
limiters. The limiters have different interfaces and need different
processing steps. Because of that, we make (rather big) changes to the
control flow in FrameCombiner. For a short while, we will mix in
deinterleaved floats when using any limiter.
NOTRY=true
Bug: webrtc:8925
Change-Id: Ie296c2b0d94f3f0078811a2a58f6fbf0f3e6e4a8
Reviewed-on: https://webrtc-review.googlesource.com/56141
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22185}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/audio_mixer/frame_combiner_unittest.cc (Browse further)
