Add aecdump support to audioproc_f.

Add a new interface to abstract away file operations. This CL temporarily
removes support for dumping the output of reverse streams. It will be easy to
restore in the new framework, although we may decide to only allow it with
the aecdump format.

We also now require the user to specify the output format, rather than
defaulting to the input format.

TEST=Bit-exact output to the previous audioproc_f version using an input wav
file, and to the legacy audioproc using an aecdump file.

Review URL: https://codereview.webrtc.org/1409943002

Cr-Commit-Position: refs/heads/master@{#10460}
10 files changed
tree: 5fb7edddeb70266ef957e540bb00c68c2d1773fd
  1. .clang-format
  2. .gitignore
  3. .gn
  4. AUTHORS
  5. BUILD.gn
  6. COPYING
  7. DEPS
  8. LICENSE
  9. LICENSE_THIRD_PARTY
  10. OWNERS
  11. PATENTS
  12. PRESUBMIT.py
  13. README.md
  14. WATCHLISTS
  15. all.gyp
  16. build_overrides/
  17. check_root_dir.py
  18. chromium/
  19. codereview.settings
  20. data/
  21. infra/
  22. license_template.txt
  23. pylintrc
  24. resources/
  25. setup_links.py
  26. sync_chromium.py
  27. talk/
  28. third_party/
  29. tools/
  30. webrtc/
README.md

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