This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
Changes in this CL:
- CaptureCursor() scans the cursor to verify that it has alpha channel.
- The AND mask of the cursor is used to reconstruct transparency if the cursor does not have alpha channel.
- CaptureCursor() always outlines the cursor when a "screen reverse" pixel detected. Previously it was only done for black and while cursors.
Added desktop_capture_unittest.MouseCursorShapeTest to test the cursor conversion code.
BUG=chromium:223147
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1627004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4210 4adac7df-926f-26a2-2b94-8c16560cd09d
Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.
BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.
R=henrika@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1628004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL eliminates the webrtc_utility_unittests test target.
NOTICE: Upon committing, this test must be removed from the
Buildbot configuration.
BUG=1843
TEST=trybots passing. Compiled and ran modules_unittests, verified the
AudioFrameOperationsTest test executes and passes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1584004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4181 4adac7df-926f-26a2-2b94-8c16560cd09d
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1460
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1595004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4165 4adac7df-926f-26a2-2b94-8c16560cd09d
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1459
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1594004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4164 4adac7df-926f-26a2-2b94-8c16560cd09d
Having this failing test being disabled in code will make it
possible to add it on the bots again, and make thus no bot
configuration update needs to be communicated when it's fixed.
BUG=1458
TEST=Compiled with GYP_DEFINES=target_arch=x64 and ran the
test successsfully on Windows. Also ran regular trybots.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1593004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4163 4adac7df-926f-26a2-2b94-8c16560cd09d
DecodedImageCallback is allowed to be called on a thread different from decoding thread. To avoid the deadlock in VCMDecodedFrameCallback::Decoded, VCMDecodedFrameCallback::_critSect should not be held while calling VCMReceiveCallback::FrameToRender.
BUG=1832
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1570004
Patch from Wu-Cheng Li <wuchengli@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4162 4adac7df-926f-26a2-2b94-8c16560cd09d
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.
BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1590004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
Reason - leading suspect of video frame corruption tracked in http://b/9216252
Note that if this turns out to not be the cause, be sure to re-revert both this change and r4145.
> Refactor jitter buffer to use separate lists for decodable and incomplete frames.
>
> This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
>
> To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
>
> This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
>
> BUG=1798
> TEST=vie_auto_test, trybots
> R=mikhal@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1522005TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1586007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4146 4adac7df-926f-26a2-2b94-8c16560cd09d
This changes the design of the jitter buffer to keeping track of decodable frames from the point when packets are inserted in the buffer, instead of searching for decodable frames when they are needed.
To accomplish this the frame_list_, which previously contained all frames (incomplete or complete, continuous or not), is split into a list of decodable_frames_ (complete, continuous) and a list of incomplete_frames_ (either incomplete or non-continuous). These frame lists are updated every time a packet is inserted.
This is another step in the direction of doing most of the work in the jitter buffer only once, when packets are inserted, instead of doing it every time we look for a frame or try to get a nack list.
BUG=1798
TEST=vie_auto_test, trybots
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1522005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4104 4adac7df-926f-26a2-2b94-8c16560cd09d
