This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.
Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
This makes it easier to refactor the interface in upcoming CLs.
Bug: None
Change-Id: I67d0216e24f087294e95ac96f7278f302bf69832
Reviewed-on: https://webrtc-review.googlesource.com/71041
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22933}
- protobuf library
- file_wrapper.h
These appear to have been left behind during the AecDump refactoring.
After this CL, APM no longer depends on zlib by default! :)
Bug: webrtc:9139
Change-Id: I12a8df2a17a575515b9c07165825f0879c4e15eb
Reviewed-on: https://webrtc-review.googlesource.com/70762
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22923}
Commit bbf21a3fd617abb37567a86e65f8ba18b8d64eb2 ("Remove dependencies on
modules:module_api from AudioProcessing") causes the build to fail with
libstdc++ due to several files using memcpy(3) or memset(3) while relying on
string.h being included implicitly by other headers.
Bug: webrtc:9139
Change-Id: Ib73284962f8694d8bed0551968265bfd13cab967
Reviewed-on: https://webrtc-review.googlesource.com/70180
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#22895}
- Directly include api/audio/audio_frame.h everywhere AudioFrame is used.
- This *will* remove transient dependencies on libjpeg and a bunch of other things from the e.g. APM.
- audio_frame.h still included from module_common_types.h for backwards compatibility with clients.
Bug: webrtc:9139, webrtc:7504
Change-Id: Id96f9268c01667fbcc29a01f5c1dd25a37836897
Reviewed-on: https://webrtc-review.googlesource.com/62464
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22845}
This CL adds tracking and reporting of packet feedback availability in
the AudioSendStream class.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I1053675d245a59c1b97fd482de88e63cbfae0038
Reviewed-on: https://webrtc-review.googlesource.com/63203
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22539}
This CL adds a boolean indicating availability of per packet feedback
to the OnAllocationLimitsChanged callback on the
BitrateAllocator::LimitObserver interface.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I5bd6e5796733da312556f2f681ff06d49ea2becc
Reviewed-on: https://webrtc-review.googlesource.com/63201
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22533}
The executable that's pushed to the device must depend on all
files that need to be on the device.
No-Try: True
TBR: phoglund@webrtc.org
Bug: chromium:755660
Change-Id: Iee041bd51e789e3ce6612fbda1582286a5cf4680
Reviewed-on: https://webrtc-review.googlesource.com/62961
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22494}
This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31
Now using rtc_source_set to be able to generate runtime deps
Original change's description:
> Split perf-test-specific resources in low_bandwidth_audio_test
>
> Bug: chromium:755660
> Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6
> Reviewed-on: https://webrtc-review.googlesource.com/61961
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22439}
No-Try: True
Bug: chromium:755660
Change-Id: I66eda6f016c98e2a8a99f230d9e0323cc09e4976
Reviewed-on: https://webrtc-review.googlesource.com/62020
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22450}
Add a flag to Android perf tests, so we can specify the number of
retries.
Bug: chromium:755660
Change-Id: Ic498373421b7e0fdf779a4659a0c79d47a59fbde
Reviewed-on: https://webrtc-review.googlesource.com/61103
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22390}
So that future CLs can de-inline its methods.
We do this by asking the caller to allocate the buffer instead of
having it as a data member.
Bug: webrtc:8982
Change-Id: I246b0973e54510fdd880c3b6875336c31334d008
Reviewed-on: https://webrtc-review.googlesource.com/60000
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22355}
The GetTransportFeedbackVector() method is only used in tests, and they
can access the class directly anyway. Keeping it is adding code bloat
and is also making upcoming refactoring more difficult.
Bug: webrtc:8975
Change-Id: I8323addb3c1461dd73b30353c8d9fe9410471c15
Reviewed-on: https://webrtc-review.googlesource.com/60860
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22349}
Just a null ID for now, but future CLs will fix that.
Bug: webrtc:8941
Change-Id: I393af0fef752ca3711421bdaf4b2e41cbe286bcf
Reviewed-on: https://webrtc-review.googlesource.com/58093
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22296}
This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.
Reason for revert: breaks downstream project
Original change's description:
> Separate test/fake_audio_device on API and implementation.
>
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
>
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
>
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}
TBR=kwiberg@webrtc.org,titovartem@webrtc.org
Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
SetTransportOverhead was used by send streams to signal the packet
overhead that they received from Call. However, call receives the value
from OnNetworkRouteChanged in WebRtcVideoChannel and
WebRtcVoiceMediaChannel which is already propagated to
RtpTransportControllerSend. By skipping the round trip, the interface on
the rtp transport controller can be reduced.
Bug: None
Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7
Reviewed-on: https://webrtc-review.googlesource.com/57182
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22226}
This is a fairly minimalistic string building class that
can be used instead of stringstream, which is discouraged
but tempting to use due to its convenient interface and
familiarity for anyone using our logging macros.
As a starter, I'm changing the string building code in
ReceiveStatisticsProxy and SendStatisticsProxy from using
stringstream and using SimpleStringBuilder instead.
In the case of SimpleStringBuilder, there's a single allocation,
it's done on the stack (fast), and minimal code is required for
each concatenation. The developer is responsible for ensuring
that the buffer size is adequate but the class won't overflow
the buffer. In dcheck-enabled builds, a check will go off if
we run out of buffer space.
As part of using SimpleStringBuilder for a small part of
rtc::LogMessage, a few more changes were made:
- SimpleStringBuilder is used for formatting errors instead of ostringstream.
- A new 'noop' state has been introduced for log messages that will be dropped.
- Use a static (singleton) noop ostream object for noop logging messages
instead of building up an actual ostringstream object that will be dropped.
- Add a LogMessageForTest class for better state inspection/testing.
- Fix benign bug in LogTest.Perf, change the test to not use File IO and
always enable it.
- Ensure that minimal work is done for noop messages.
- Remove dependency on rtc::Thread.
- Add tests for the extra_ field, correctly parsed paths and noop handling.
Bug: webrtc:8529, webrtc:4364, webrtc:8933
Change-Id: Ifa258c135135945e4560d9e24315f7d96f784acb
Reviewed-on: https://webrtc-review.googlesource.com/55520
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22203}
Using a mock of rtp transport controller send in audio send stream unit
tests. This reduces the dependencies and makes the tests more focused
on testing the functionality of audio send stream itself.
Bug: webrtc:8415
Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2
Reviewed-on: https://webrtc-review.googlesource.com/56600
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22161}
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.
With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.
Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
Since rtp transport controller send owns the congestion controller it
also should own the bitrate configuration logic, this way it can
initialize the send side congestion controller with the bitrate
configuration.
Bug: webrtc:8415
Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553
Reviewed-on: https://webrtc-review.googlesource.com/54303
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22127}
The audio send stream unit tests did not use the mocks injected to the
fake rtp transport controller send. This CL prepares for removing the
fake controller which makes it harder to refactor the rtp transport
controller interface.
Bug: webrt:8415
Change-Id: I73f7d105e66f9beb80aeaa92f3490cd61c80c5b8
Reviewed-on: https://webrtc-review.googlesource.com/54302
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22102}
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.
Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
