Added WebRTC audio quality test using fake device; improved fake device
commitb0a18bb4d74babb14f98808f0270c540cc2efd13
authorphoglund <phoglund@chromium.org>
Fri, 9 Jan 2015 08:14:18 +0000 (9 00:14 -0800)
committerCommit bot <commit-bot@chromium.org>
Fri, 9 Jan 2015 08:15:08 +0000 (9 08:15 +0000)
treed22061aaa7d857646cab269ba511e4f4b5fe521a
parentffb643afc26c5259015c0d4d2fc1254cffd0bc78
Added WebRTC audio quality test using fake device; improved fake device

This finally adds a end-to-end WebRTC audio quality test which includes
getUserMedia and the capture path (except, obviously, the os-specific
code which actually interacts with the capture device etc).

This patch also makes the audio input device generic. It will now
convert its input .wav file to suit the audio bus. I had to do this
because the PESQ tests require files to be mono, 16-bit and so on.

BUG=446859

Review URL: https://codereview.chromium.org/831693004

Cr-Commit-Position: refs/heads/master@{#310736}
chrome/browser/media/chrome_webrtc_audio_quality_browsertest.cc
chrome/test/data/webrtc/resources/human-voice-linux.wav.sha1
chrome/test/data/webrtc/resources/human-voice-mac.wav.sha1
chrome/test/data/webrtc/resources/human-voice-win.wav.sha1
media/audio/fake_audio_input_stream.cc
media/audio/fake_audio_input_stream.h
media/base/media_switches.cc