1295561 - [WebRTC] Poor outbound audio quality in FF 49 beta 3 using opus codec

Status: RESOLVED DUPLICATE of bug 1297083

(Core :: WebRTC: Audio/Video, defect)

Description

[marko.seidenglanz]

Attachment: SDP.txt

User Agent: Mozilla/5.0 (X11; Linux x86_64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/52.0.2743.116 Safari/537.36

Steps to reproduce:

Created Connection between Firefox and FreeSWITCH.


Actual results:

Bad outbound audio quality (FF -> Freeswitch). The other direction works as expected. The problem only was noticed since FF 49 beta. All versions before worked as expected. Does it have something todo with the removal of 'Hello' in FF 49?

Comment 1

[marko.seidenglanz]

Hello.

We investigate the same problem when switching to G711, so it does not seem to be codec problem. 
We also tested via https://opentokrtc.com/ with the same results, so it does not seem to be a problem with Freeswitch.

Beta 1 was ok.
Beta 4 has the same issue.

kind regards,
Marko

Comment 2

[marko.seidenglanz]

Same problems with https://demo.easyrtc.com/demos/demo_audio_only.html

Comment 3

[Randell Jesup [:jesup] (needinfo me)]

Thanks for the report - can we get more info?

I tested it in beta4 on mac with someone else with easyrtc, worked fine (and worked fine with full_duplex on and off).  I also tried Windows (running Nightly).  I also reviewed all modifications to Beta to see if anything might have caused it; didn't see anything suspicious 

What OS/version were you on?  Have you tried Aurora (Developer Edition) or Nightly? (could you?)

What microphone were you using?  What type of connection?  (USB headset, 3.5mm headset, built-in mic/speakers, etc?)

Can you characterize (or record) the "Bad outbound quality"?  (Even a rough recording would help).

Can you get logs:  run with env vars NSPR_LOG_MODULES=MediaManager:4,GetUserMedia:4

Were there any other tabs actively capturing audio, or just the one?  Does it sound ok using https://mozilla.github.io/webrtc-landing/gum_test.html (Audio or Audio & Video)?  (Headset suggested, and it may "squelch" some due to the AEC, since the AEC can decide your voice is an echo of your voice).

Thanks!  Since you're seeing this in beta, we want to resolve it ASAP.

Comment 4

[marko.seidenglanz]

Hello.

I've tried Aurora / Nightly and observed the same problems.
We've used USB headsets (Plantronics) and also 3.5mm.

Using https://mozilla.github.io/webrtc-landing/gum_test.html works and has good voice quality. As you wrote, the audio get's clipped sometimes, but there is no distortion.

We made several more tests on Linux and Windows machines and in a fraction (20%) of all testruns it is working correctly. When the quality was good and we did some more calls within the same browser session it worked. After restarting the browser the problem appeared again.

I've also disabled aec and full-duplex mode, but it had no effects.

An audio recording and NSPR Logs were attached.

Comment 5

[marko.seidenglanz]

Attachment: testrecording.mp3

Audio recording (channel FF -> Freeswitch (distorded voice) first 15 seconds / channel Freeswitch -> FF in second 15 seconds). The audio was grabbed on Freeswitch.

Comment 6

[marko.seidenglanz]

Attachment: NSPR Log

Comment 7

[marko.seidenglanz]

We've tried several Headsets (USB and 3.5mm) with no better results.

Soundcard: HDA Intel PCH, ALC887-VD Analog.

Comment 8

[marko.seidenglanz]

We also changed to the new Stream API and use addTrack instead of addStream but it had no effect.

Comment 9

[marko.seidenglanz]

Here are some RTP Stats, if that helps anything:

RTP-Statistiken
outbound_rtcp_audio_0

Lokal: 17:30:24 GMT+0200 inboundrtp SSRC: 2767720019 Empfangen: 7942 Pakete (1240.94 Kb) Verloren: 1 Jitter: 0.081 RTT: 1 ms
inbound_rtp_audio_0

Jitter-Puffer-Verzögerung: 61 ms

Lokal: 17:30:29 GMT+0200 inboundrtp SSRC: 3016340397 Empfangen: 4263 Pakete (716.05 Kb) Verloren: 0 Jitter: 0.02

Extern: 17:30:25 GMT+0200 outboundrtp SSRC: 3016340397 Gesendet: 4028 Pakete (668.71 Kb)
outbound_rtp_audio_0

Lokal: 17:30:29 GMT+0200 outboundrtp SSRC: 2767720019 Gesendet: 8593 Pakete (1527.27 Kb)

Extern: 17:30:24 GMT+0200 inboundrtp SSRC: 2767720019 Empfangen: 7942 Pakete (1240.94 Kb) Verloren: 1 Jitter: 0.081 RTT: 1 ms
inbound_rtcp_audio_0

Lokal: 17:30:25 GMT+0200 outboundrtp SSRC: 3016340397 Gesendet: 4028 Pakete (668.71 Kb)

Comment 10

[Randell Jesup [:jesup] (needinfo me)]

Can you retry this with tomorrow's nightly (2016-08-24 build)?  We landed a fix (soon to be uplifted) which would cause audio to inserted N times if there are N GetUserMedia streams active for that mic.  (bug 1297083).  This would affect 49 and newer

Comment 11

[marko.seidenglanz]

That was exactly the problem. We grabbed mic audio several times. No problems with 2016-08-24 build.

Thank you very much!!!