brandtr b2def1d06f Add batch mode to VideoProcessor integration tests.
Prior to this CL, the encoding/decoding in the VideoProcessor integration
tests were run "online", in the sense that rate allocations could be
changed in between frames. This is useful for evaluating the rate control
of SW codecs, which is one of the reasons for the existence of these
integration tests in the first place.

This CL adds a batch mode, in which the tests are run "offline". The two
main differences to the original mode are: 1) rate control metrics are
calculated after the fact, and 2) no rate allocation changes are allowed
during the test. Difference 1) is the reason for this CL, as HW codecs
that are pipelining will not work well when rate control metrics are
calculated right after a frame has been sent for encode. Difference 2)
is a side effect of the introduction of the batch mode. If we want to
be able to support online rate allocation for pipelining HW codecs in
the future, this can be introduced by adding a delay between encoding
and rate allocation. This was not deemed necessary at this point in time,
and hence this CL does not do that.

The batch mode is only intended to be used for manual experimentation
on devices with HW codecs, and the integration tests running on the
bots should thus NOT use batch mode.

BUG=webrtc:6634

Review-Url: https://codereview.webrtc.org/2707023008
Cr-Commit-Position: refs/heads/master@{#17164}
2017-03-10 12:20:10 +00:00
.gn
2017-03-08 10:12:11 +00:00
2017-03-07 00:42:19 +00:00
2017-03-09 10:04:57 +00:00
2017-01-20 20:45:07 +00:00
2015-09-11 09:04:09 +00:00

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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