[One (M7) Original Android Development] dyndragon:
I have reported this in the past (see [MENTION=1815755]intervigil[/MENTION] 's response at http://xdaforums.com/showpost.php?p=43370046&postcount=2299) . There are audio routing issues that are still unresolved with apps that provide voice or video conference/chat or recording. [MENTION=4635674]n3ocort3x[/MENTION] if you could figure it out, I would love you forever. This is my one remaining problem that sometimes makes me boot up in Sense when I need to use Google Hangouts video or audio conferencing. I tried tracking down the audio usecase that is missing (if it is actually that) but it is beyond my ability.In Google Hangouts, specifically you get no audio out of the speakers, but if you switch the audio to the "handset" you get audio. Logs attached of a video call where I switch audio modes between speaker and handset. I think of interest:

D/alsa_ucm( 235): snd_use_case_set(): uc_mgr 0x41f88bc8 identifier _verb value Inactive

D/alsa_ucm( 235): Set mixer controls for Voice Call IP enable 0

D/alsa_ucm( 235): Setting mixer control: SLIM_0_RX_Voice Mixer Voip, value: 0

D/alsa_ucm( 235): Setting mixer control: Voip_Tx Mixer SLIM_0_TX_Voip, value: 0

D/alsa_ucm( 235): snd_use_case_set(): uc_mgr 0x41f88bc8 identifier _disdev value VOIP Line

D/alsa_ucm( 235): Set mixer controls for VOIP Line enable 0

D/alsa_ucm( 235): Setting mixer control: SLIM TX7 MUX, value: ZERO

D/alsa_ucm( 235): Setting mixer control: DEC9 MUX, value: ZERO

D/alsa_ucm( 235): Setting multi value: DEC9 Volume

D/alsa_ucm( 235): Setting mixer control: MICBIAS3 CAPLESS Switch, value: 0

D/alsa_ucm( 235): snd_use_case_set(): uc_mgr 0x41f88bc8 identifier _disdev value VOIP Speaker

D/alsa_ucm( 235): Set mixer controls for VOIP Speaker enable 0

D/alsa_ucm( 235): Setting mixer control: MI2S_RX_Voice Mixer Voip, value: 0

W/libjingle( 1528): Warning(webrtcvoiceengine.cc:1097): WebRtc:StatisticsRTP() no statistics available

W/libjingle( 1528): Warning(webrtcvoiceengine.cc:1097): WebRtc:SenderInfoReceived No received SR

W/libjingle( 1528): Warning(webrtcvoiceengine.cc:1097): WebRtc:StatisticsRTP() no statistics available

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

D/ALSADevice( 235): setVoipVolume: volume 100

D/ALSADevice( 235): setMixerControl:: name Voip Rx Volume value 100 index 0

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

E/AudioPolicyManagerALSA( 235): unknown stream type

---------- Post added at 12:40 PM ---------- Previous post was at 12:29 PM ----------

Also, something dumb is happening and it might be me. I usually am able to view logcats while on my device with alogcat, but I've just recently noticed while trying to get logs for this audio problem with Hangouts that I don't have any logs in alogcat. Did I disable logging somehow or is logging just not turned on yet?