Chrome version: 50.0.2661.37 / 7978.20.0 dev
Device: Sentry

Steps to reproduce:

1. Browse to https://webrtc.github.io/samples/src/content/peerconnection/audio/
2. Open the javascript console (alt+cmd+j or ctr+shift+j) and watch for any errors
3. For each codec in the dropdown menu and Press 'Call'
4. Observe the bitrate graphs for each codec: 
a. For 'Opus' codec, the Bitrate graph shows values ranging between 10 and 30 kbps (not in the expected range of ~40kbps) 
b. For 'iSAC 16k' codec, the Bitrate graph shows values ranging between 10 and 15 kbps (not in the expected range of ~30 kbps)
c. For G722 codec, the Bitrate graph shows values ranging around 70kbps d. For PCMU codec, the bitrate graph shows values ranging around 70kbps
5. Observe that the Packets sent per second graph shows nonzero values
(working as expected)

Expected results:
Opus:
Bitrate range: ~40 kbps
Packets sent per second range: ~50 packets

iSAC 16k
Bitrate range: ~30 kbps
Packets sent per second range: ~35 packets

G722
Bitrate range: ~70 kbps
Packets sent per second range: ~50 packets

PCMU
Bitrate range: ~70 kbps
Packets sent per second range: ~50 packets

Actual results:
Opus:
Bitrate range: 10-30 kbps 
Packets sent per second range: ~50 packets

iSAC 16k
Bitrate range: 10-15kbps
Packets sent per second range: ~35 packets

G722
Bitrate range: ~70 kbps
Packets sent per second range: ~50 packets

PCMU
Bitrate range: ~70 kbps
Packets sent per second range: ~50 packets

 

Deleted: Device_sentry_webrtc_internals_dump.txt 131 KB

Device_sentry_webrtc_internals_dump.txt 131 KB View Download