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}
If the tests are run in a different order, the test might fail.
We fix this by resetting the histogram data at the start of the test.
Change-Id: I6fb349609842b55f416cf2ec8cd93d0b4328960e
Bug: chromium:1430806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323801
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Fredrik Hernqvist <fhernqvist@google.com>
Cr-Commit-Position: refs/heads/main@{#40946}
Removes the top 3 filtering based on frame energy. This behaviour is
unexpected for many application developers and the platform should not
have such arbitrary limitations. Developers can still implement top-N
filtering using WebAudio or an SFU (recommended to increase
scalability).
Performance is not really a concern in this case since decoders on all
receive streams are called regardless if they are mixed or not
(assuming packets are received).
This also fixes glitches caused by the current implementation since
sources are not ramped out.
Bug: chromium:1446655,webrtc:13818
Change-Id: I179a6d68d2517b94ff2d99ec269031a54e5099e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310180
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40349}
This adds the histogram WebRTC.Audio.AudioMixer.NewHighestSourceCount
which logs the highest number of sources an AudioMixer has had. The
statistic is logged whenever the highest number of sources increases.
This allows us to differentiate the statistic to see how many times
the mixer has had a certain maximum number of sources.
Chromium CL:
https://chromium-review.googlesource.com/c/chromium/src/+/4414896
Bug: chromium:1430806
Change-Id: Iab92e201a0b667741cc8f3bbbed92fa989d7fcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300860
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39833}
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}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
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}
This allows mixing different number of streams depending on the
client's capabilities.
This CL adds `WebRTC.Audio.AudioMixer.NumIncomingActiveStreams2`,
which is defined in [1], since the histogram is not logged anymore
as enum.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/2883627
Bug: webrtc:12746
Change-Id: I0d9b3888f0f95269806539e33b56619b757a5c68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218160
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Cr-Commit-Position: refs/heads/master@{#34024}
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.
Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
To remove global task factory, rtc::TaskQueue need to loose it's convenient constructor
TaskQueueForTest can be used instead in tests and keep the convenient constructor.
Also cleanup the TaskQueueForTest a bit:
move the class to webrtc namespace
add default constructor
disallow copy using language construct instead of macro
cleanup build dependencies
rename build target (to match move out of the rtc namespace)
Bug: webrtc:10284
Change-Id: I17fddf3f8d4f363df7d495c28a5b0a28abda1ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127571
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27193}
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}
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}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This eliminates a lot of warnings when running AudioMixer unittests.
Bug: None
Change-Id: I978e859e8904bb382843a9c23a35ace1884aa4b4
Reviewed-on: https://webrtc-review.googlesource.com/58981
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22245}
This class adds a convenience method that allows *sending* a task
to the queue (as opposed to posting). Sending is essentially
Post+Wait, a pattern that we don't want to encourage use of
in production code, but is convenient to have from a testing
perspective and there are already several places in the
source code where we use it.
Change-Id: I6efd1b2257e6c641294bb6e4eb53b0021d9553ca
Bug: webrtc:8848
Reviewed-on: https://webrtc-review.googlesource.com/50441
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22022}
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/audio_mixer_impl_unittest.cc (Browse further)
