This is to avoid using 0 as a default value.
Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp.
Bug: webrtc:13609
Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118
Reviewed-by: Olov Brändström <brandstrom@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39177}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
This CL removes PushCaptureData(), which is unused.
The reason I'm removing it is since this method is cauing chromium-style
violations for all files that includes
modules/audio_device/include/audio_device_defines.h, and it's annoying
to suppress it everywhere.
Bug: webrtc:8659
Change-Id: I9133d05259075d8e8ec89b764be934f37b5fa77e
Reviewed-on: https://webrtc-review.googlesource.com/66404
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22717}
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_device/include/mock_audio_transport.h (Browse further)
