Erik Språng 08127a9449 Reland #2 of Issue 2434073003: Extract bitrate allocation ...
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:

1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.

Please review only the changes after patch set 1.

Original description:

Extract bitrate allocation of spatial/temporal layers out of codec impl.

This CL makes a number of intervowen changes:

* Add BitrateAllocation struct, that contains a codec independent view
  of how the target bitrate is distributed over spatial and temporal
  layers.

* Adds the BitrateAllocator interface, which takes a bitrate and frame
  rate and produces a BitrateAllocation.

* A default (non layered) implementation is added, and
  SimulcastRateAllocator is extended to fully handle VP8 allocation.
  This includes capturing TemporalLayer instances created by the
  encoder.

* ViEEncoder now owns both the bitrate allocator and the temporal layer
  factories for VP8. This allows allocation to happen fully outside of
  the encoder implementation.

This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.

BUG=webrtc:6301
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2510583002 .

Cr-Commit-Position: refs/heads/master@{#15105}
2016-11-16 15:41:45 +00:00
2016-11-16 11:09:07 +00:00
2016-11-14 13:54:29 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +00:00
2016-10-19 11:52:19 +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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