Database Users

[DEV] Android 4.0.1 Audio problems

HD2 audio dev.
Good news - Nexus1:
<rapmv78> I got audio partially working on nexus one
<cruonit> @rapmv78 how did you do it... outline
<cruonit> need it for hd2 :|
<rapmv78> The issue seems to be that the audio legacymode driver which we have does not expect multiple threads of audiopolicy services especially when there is request from each thread to set format/channels/sampling rate etc. There is code in openOutputStream in AudioHardware.cpp, which seems to be the issue. I have a hack which just ignores different set requests. Only first one sticks. The following code modification seems to
<rapmv78> if (mOutput) { if (status) { if (0) { *status = INVALID_OPERATION; } else { LOGD("Output stream already exists, using existing channels and format"); *status = NO_ERROR; } } return mOutput; }
<rapmv78> I tried it with the music app and was able to play mp3s. However there is a small issue with the audio switching from speaker to headset randomly. There are still some issues.
<rapmv78> just build audio.primary.qsd8k.so with the hack and drop it in /system/lib/hw and restart android
other discussion references:
http://forum.xda-developers.com/showthread.php?t=1348670&page=6
http://forum.xda-developers.com/showthread.php?t=1348479&page=3
already started topics:
http://forum.xda-developers.com/showthread.php?t=793353

now does this hack work as far as the incall audio, voice output and, Phone sounds ECT.? Or just for what you have stated?

some ringtones work some not :|

Cruonit said:
HD2 audio dev.
Good news - Nexus1:
<rapmv78> I got audio partially working on nexus one
<cruonit> @rapmv78 how did you do it... outline
<cruonit> need it for hd2 :|
<rapmv78> The issue seems to be that the audio legacymode driver which we have does not expect multiple threads of audiopolicy services especially when there is request from each thread to set format/channels/sampling rate etc. There is code in openOutputStream in AudioHardware.cpp, which seems to be the issue. I have a hack which just ignores different set requests. Only first one sticks. The following code modification seems to
<rapmv78> if (mOutput) { if (status) { if (0) { *status = INVALID_OPERATION; } else { LOGD("Output stream already exists, using existing channels and format"); *status = NO_ERROR; } } return mOutput; }
<rapmv78> I tried it with the music app and was able to play mp3s. However there is a small issue with the audio switching from speaker to headset randomly. There are still some issues.
<rapmv78> just build audio.primary.qsd8k.so with the hack and drop it in /system/lib/hw and restart android
other discussion references:
http://forum.xda-developers.com/showthread.php?t=1348670&page=6
http://forum.xda-developers.com/showthread.php?t=1348479&page=3
already started topics:
http://forum.xda-developers.com/showthread.php?t=793353
Click to expand...
Click to collapse
Post this in the Dev section I think, this is definitely worthy of more attention. Note that Tytung has thanked your post

ALSA on the Nexus 4 (temporary volume control)

Please remember to make a backup image of your phone before experimenting.
============ WARNING!! =============
The highest setting is VERY loud (as I found out) and could potentially rip apart the speaker on your phone!
Exercise extreme caution when using this. You risk permanent damage to your phone and your ears.​
2013-Jan-20
-----------------
Thanks xda-developers members "skvalex" who wrote ALSAMixer and "chdlock" who is helping me figure out
how ALSA is mapping the audio chipset, we can now temporarily adjust the Digital Volume on the Nexus 4.
Unfortunately, the selected volume doesn't "stick" because the audio driver re-writes it every time it re-opens an audio stream.
A stream is usually opened when a music track begins to play so then the digital volume will reset back to default.
We need a kernel developer to modify the audio driver so it opens an audio stream using a user-selected Volume level.
***
If you don't mind setting the volume frequently or just want to try this out here's what you need:
1) Get ALSAMixer from the Play Store. It was updated recently and you need the latest version.
Please consider a donation to the author.
2) You can re-program the audio chipset by editing fields in ALSAMixer. Here is a list of what we can control so far.
Headphones volume:
Left: RX1 Digital Volume (numid=27)
Right: RX2 Digital Volume (numid=28)
Speaker volume:
RX3 Digital volume (numid=29)
Camcorder volume:
DEC6 Volume (numid=39)
Microphone volume:
DEC7 Volume (numid=40)
-----------------------------------------------
old OPs
-----------------------------------------------
2013-Jan-15
----------------
The creator of ALSAMixer updated it to correctly install for Nexus4. (Uninstall removes the app but the ALSA libraries remain on the phone.)
https://play.google.com/store/apps/details?id=com.skvalex.alsamixer&hl=en
Thank you so much!
--------
http://forum.xda-developers.com/attachment.php?attachmentid=1653124&d=1358438577
This is a piped output of "alsa_amixer -c 0 contents" from a stock Nexus 4 (4.2.1)
2013-Jan-13
-----------------
Is anyone successfully running ALSA on their Nexus 4?
If you do please share instructions to do so.
I am on stock 4.2.1(rooted/busybox) and I ran the AlsaMixer from
the Play Store but it doesn't seem to complete the installation.

http://forum.xda-developers.com/sho...ited Edition Kit★☆★ | aokp,minco,pa,rasbean |
Shameless promotion
Sent from my Nexus 4

Next you're going to tell me that you want systemd and PulseAudio on your device too, right?

I wiped to stock,
I installed AOKP-jb-mr1-build1,
then Nocturnal 1.4 LE for AOKP
and the ALSAMixer still doesn't work. It installs but I get the same blank screen with no settings.
What am I doing wrong?

I'm not sure what that app is, I just meant that some of the audio mods in the kit ROM referred to ALSA audio libs.
As you can tell I'm not the expert, sorry if that's not what you were referring too
Sent from my Nexus 4

I am looking for a way to control the audio stream parameters in particular the gain of the microphone.
Somebody suggested using ALSA and I'm trying get it going.

-Mindroid- said:
I am looking for a way to control the audio stream parameters in particular the gain of the microphone.
Somebody suggested using ALSA and I'm trying get it going.
Click to expand...
Click to collapse
I have only a Nexus 7, and the SoC is different from the Nexus 4, but look for alsa_amixer (installed by AlsaMixer) from within a console and execute
"alsa_amixer -c X contents" (without the quotes)
where X is one of the card numbers shown in your /proc/asound directory (e.g. card0 [X=0], card1 [X=1], etc). In my case, X=1 is the actual card where (a lot of) values can be set. One of those values is called "ADC Boost Gain"(I guess a programmable analog gain?).
I established a call with CSipSimple and successfully raised that parameter by 24dB in mid-call by executing "alsa_amixer -c 1 cset numid=20 2" [the default value was 0 and each step increases the level by 12dB]
You can query individual parameters by executing alsa_amixer -c X cget numid=Y
YMMV with the Nexus 4. Good luck!

Thank you! This looks really promising.
I only have "card0" aka [apq8064tablasnd].
When trying to execute "alsa_mixer -c 0 contents" I get an error stating that I'm missing
library "libasound.so". Any ideas what could be wrong?

I'm not sure whether libasound.so came with the AlsaMixer app. If it did you may need to copy the library into a system library path, e.g. /system/lib. If not, I could send you the library that I have. I actually forgot whether I found the library somewhere online or whether I cross-compiled it myself...
Sent from my Nexus 7 using xda app-developers app

ALSA on Nexus 4 sound capabilities
ALSAMixer in the Play Store was updated to support Nexus 4. See OP.
---------------------------------------------------------------
The attached file is the output of running
alsa_amixer -c 0
I'm new to ALSA and have trouble figuring out which one is the microphone gain control.

I have installed this app but unfortunately I don't know how to completely uninstall it help please?:thumbup:
..sent from space.

amarb70 said:
I have installed this app but unfortunately I don't know how to completely uninstall it help please?:thumbup:
..sent from space.
Click to expand...
Click to collapse
If I understand correctly the app is just an installer that puts the ALSA libraries on your system.
If you are not using them it probably doesn't matter if they are there or not. You can contact the app creator about it.

-Mindroid- said:
ALSAMixer in the Play Store was updated to support Nexus 4. See OP.
---------------------------------------------------------------
The attached file is the output of running
alsa_amixer -c 0
I'm new to ALSA and have trouble figuring out which one is the microphone gain control.
Click to expand...
Click to collapse
Post the output of
alsa_amixer -c 0 contents
Any labels containing "ADC gain" or "Boost" would be prime candiates

amarb70 said:
I have installed this app but unfortunately I don't know how to completely uninstall it help please?:thumbup:
..sent from space.
Click to expand...
Click to collapse
Restore your back up
Sent from my Nexus 4 using xda premium

alsa_amixer c- 0 contents
chdloc said:
Post the output of
alsa_amixer -c 0 contents
Any labels containing "ADC gain" or "Boost" would be prime candiates
Click to expand...
Click to collapse
Thank you for being so patient with me!
I'm such a Linux noob. It took me forever to figure out that I need 'su' to do anything.
I will update the OP to link to this file.

-Mindroid- said:
Thank you for being so patient with me!
I'm such a Linux noob. It took me forever to figure out that I need 'su' to do anything.
I will update the OP to link to this file.
Click to expand...
Click to collapse
Holy smokes! This is one complicated mixer! And I've seen many.
It may help in narrowing choices down by you capturing another
alsa_amixer -c 0 contents
while you are on a call and look at the diffs. While you do that (being on a call, but don't run anything else that may use audio such as music playback apps) also report the output of
cat /proc/asound/card0/pcm0p/sub0/status
to make sure that phone calls are actually routed through ALSA. Anything but "CLOSED" is good.
I do not have a Nexus 4 so I cannot help trying to find the best parameters for a given application. The app ALSAMixer should help I would think.
Does this program now actually display anything?

chdloc said:
Holy smokes! This is one complicated mixer! And I've seen many.
It may help in narrowing choices down by you capturing another
alsa_amixer -c 0 contents
while you are on a call and look at the diffs. While you do that (being on a call, but don't run anything else that may use audio such as music playback apps) also report the output of
cat /proc/asound/card0/pcm0p/sub0/status
to make sure that phone calls are actually routed through ALSA. Anything but "CLOSED" is good.
I do not have a Nexus 4 so I cannot help trying to find the best parameters for a given application. The app ALSAMixer should help I would think.
Does this program now actually display anything?
Click to expand...
Click to collapse
It's a long shot but if you are in the New York tri-state area I wouldn't mind meeting up and letting you mess around with the Nexus4.
The invitation is open to anyone with ALSA experience. I would love to learn more about it first-hand.
As far as I can tell:
- ALSAMixer installs the ALSA libraries and displays the control set. I tried modifying a value with it and it didn't work.
It's a matter of wrong syntax I think, I will contact the app creator about it.
- Terminal Emulator works. I was able to change the value of "AUX_PGA_LEFT Volume" with the following command:
alsa_amixer cset numid=54 39
I'm not sure what this one does yet but I will investigate later today.
- unfortunately, during a phone call "proc/asound/card0/pcm0p/sub0/status" (playback) as well as
"proc/asound/card0/pcm0c/sub0/status" (capture) report "closed". Earlier I tried recording a voice call from an app
and trying to prepare an audio stream with MediaRecorder.AudioSource.VOICE_CALL throws an exception.
- when recording an audio stream with MediaRecorder.AudioSource.MIC and CAMCORDER
"proc/asound/card0/pcm0c/sub0/status" reports audio stream parameters. I'm attaching the piped output of
"alsa_amixer -c 0 contents" during a voice call, recording with mic and recording with camcorder.

-Mindroid- said:
[...]
- when recording an audio stream with MediaRecorder.AudioSource.MIC and CAMCORDER
"proc/asound/card0/pcm0c/sub0/status" reports audio stream parameters. I'm attaching the piped output of
"alsa_amixer -c 0 contents" during a voice call, recording with mic and recording with camcorder.
Click to expand...
Click to collapse
This is fun...
There is certainly a difference between the "idle" set of parameters you had posted earlier and the ones that you took during a voice call.
I found some good information that explains the parameters here
https://github.com/psykick5/android_device_lge_mako/blob/master/snd_soc_msm_2x_Fusion3
Try to change numid=48 (i.e. 'ADC1 Volume') and numid=40 (i.e. 'DEC7 Volume'); these two parameters should change the microphone level, provided automatic gain control does not compensate for the changes.
(It bothers me, though, that the range of values shown by alsa_amixer and the ones shown in the above link don't match)
Make sure that the changes stick by issuing
alsa_amixer cget numid=X
For phone calls I would certainly change numid=65 (i.e. 'TX6 HPF cut off') from 0 (apparently really only a DC offset filter) to 2 (apparently a high-pass filter with a cutoff frequency of 150Hz), which would be good enough for HD voice.

chdloc said:
This is fun...
There is certainly a difference between the "idle" set of parameters you had posted earlier and the ones that you took during a voice call.
I found some good information that explains the parameters here
https://github.com/psykick5/android_device_lge_mako/blob/master/snd_soc_msm_2x_Fusion3
Click to expand...
Click to collapse
Great find! This seems to be exactly what we need.
Most intriguing there are also facilities for Voice Call and FM Radio audio streams. Those are not currently working on
the Nexus4 at the moment and this might shed some light on a possible solution.
I find some of their nomenclature confusing. They use "TX" for recording and "RX" for playback.
Also they seem to set values twice - :1:0 for turn off and :1:1 for turn on. Do I have to issue commands twice?
I confirmed with 'cget' that I am indeed setting values with 'cset'.
chdloc said:
Try to change numid=48 (i.e. 'ADC1 Volume') and numid=40 (i.e. 'DEC7 Volume'); these two parameters should change the microphone level, provided automatic gain control does not compensate for the changes.
(It bothers me, though, that the range of values shown by alsa_amixer and the ones shown in the above link don't match)
Make sure that the changes stick by issuing
alsa_amixer cget numid=X
Click to expand...
Click to collapse
It seems that DEC7 controls to the MICROPHONE stream and DEC6 the CAMCORDER.
I will focus my attention on these.
chdloc said:
For phone calls I would certainly change numid=65 (i.e. 'TX6 HPF cut off') from 0 (apparently really only a DC offset filter) to 2 (apparently a high-pass filter with a cutoff frequency of 150Hz), which would be good enough for HD voice.
Click to expand...
Click to collapse
I was wondering about that myself. There is very bad low frequency noise on video recordings so I will
definetly look into this.

-Mindroid- said:
I find some of their nomenclature confusing. They use "TX" for recording and "RX" for playback.
Click to expand...
Click to collapse
Actually, the nomenclature is quite common, TX, i.e. transmit [to the network] denotes the signal to be transmitted (taken from from microphone)
while RX is the receive signal [from the network] to be played back through a loudspeaker.
-Mindroid- said:
Also they seem to set values twice - :1:0 for turn off and :1:1 for turn on. Do I have to issue commands twice?
Click to expand...
Click to collapse
No, you will have to issue the command only once. Without having looked at the underlying code I tend to think that the first number denotes the type, i.e. integer vs string, and the second number the actual value.
Another interesting observation is that the SoC of the Nexus 4 seems to support SRS TruMedia via an effects mixer, which reportedly has potential to improve audio considerably (http://www.srstechnologies.com/content.aspx?id=1269 , I believe).

ALSA from chroot -- Audio for the other side

First, a short story
After installing a chroot debian on my s2 I noticed that vlc player isn't working.
Sadly it took me several weeks to notice that sound in general wasn't working at all.
Problem: There was simply nowhere to output the sound.
So I tried to get alsa working from chroot to get some form of audio output.
and here it is.
ALSA ON CHROOT LINUX
The bumpy road to half broken (and therefore other half working) linux audio​
Once you have a working chroot (with correct mounts that is), download and install the following packages:
Code:
alsa-base alsa-oss alsa-utils
Now run
Code:
alsamixer
to open the alsa sound mixer.
There you should unmute
Code:
DL1 MM_EXT DL1 Mixer Multimedia DL1 PDM
Then set
Code:
HS Left
and
Code:
HS Right
to
Code:
HS DAC
This should enable headset output
ALso unmute
Code:
Earphone
to well....use the Earphone speaker.
To enable the Main speaker, unmute
Code:
DL2 Mixer Multimedia DL2 Mono
And set
Code:
HF Right
and
Code:
HF Left
to
Code:
HF DAC
For other devices, this section will be different. Just look through alsamixer and try to make sense of the countless options there.
Finished more or less.
Now you can play audio from chroot.:victory:
Notice: If you run any sound from android or plug the headset in/out, you need to reapply these settings.
To avoid that I run a script that continously re-enables the mentioned settings. (mainly the headset)
It looks something like this
Code:
#!/bin/bash
while true; do
[INDENT]
amixer -c 0 csent numid=74 1
amixer -c 0 csent numid=73 1
amixer -c 0 csent numid=35 1
amixer -c 0 csent numid=52 1
amixer -c 0 csent numid=33 1
sleep 0.5
[/INDENT]
done
This probably isn't the best way of doing it (it might actually be the worst), but it works.
It might put load on the cpu or it might not, but for me, running this decreases my SoD frequency. (idek why)
Also the best way to run the script is a few seconds after the system started using the "nohup" command. That way you can run it and close the terminal and it will still be active. To kill it, search its pid using
Code:
ps -x
and kill it with
Code:
kill (insert pid here)
If you want you can also run it as a start up script.
Though if you do that, you might see that it runs but doesn't do anything.
Don't panic. Or do if you want to.
But just restart the script anyways.
OPTIONAL: (because I force you to use the stuff above)
We can play no more than ONE audio source at a time
Soloution: PulseAudio
(I will fill this shortly....)
(there is also MPD which works...)
So for now we have
Solution 2: Dmixer
Dmixer is what it name sounds like. It downmixes multiple sources into a single audio stream.
And it works (sometimes).
So here is how to.
Simply copy the content of the file below into a new file called .asoundrc in /home/username
Code:
pcm.!default {
type plug
slave.pcm "dmixer"
}
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
This one is mainly pulled directly from the alsa project site and should work most of the time.
hw:0,0 might need changing on different devices. Same with card 0.
run
Code:
aplay -l
and use that to find the correct card and device.
???
The standard: I am not responsible for anything that happens to you, your phone or anything else.
Also I didn't invent any of this stuff myself. All credit goes to the original developers.
I just wrote this guide (if you can call it that)
Oh...and if you can't understand my english, well.... I'm sorry but i can't do better.

ruleh said:
First, a short story
After installing a chroot debian on my s2 I noticed that vlc player isn't working.
Sadly it took me several weeks to notice that sound in general wasn't working at all.
Problem: There was simply nowhere to output the sound.
So I tried to get alsa working from chroot to get some form of audio output.
and here it is.
ALSA ON CHROOT LINUX
The bumpy road to half broken (and therefore other half working) linux audio​
Once you have a working chroot (with correct mounts that is), download and install the following packages:
Code:
alsa-base alsa-oss alsa-utils
Now run
Code:
alsamixer
to open the alsa sound mixer.
There you should unmute
Code:
DL1 MM_EXT DL1 Mixer Multimedia DL1 PDM
Then set
Code:
HS Left
and
Code:
HS Right
to
Code:
HS DAC
This should enable headset output
ALso unmute
Code:
Earphone
to well....use the Earphone speaker.
To enable the Main speaker, unmute
Code:
DL2 Mixer Multimedia DL2 Mono
And set
Code:
HF Right
and
Code:
HF Left
to
Code:
HF DAC
For other devices, this section will be different. Just look through alsamixer and try to make sense of the countless options there.
Finished more or less.
Now you can play audio from chroot.:victory:
Notice: If you run any sound from android or plug the headset in/out, you need to reapply these settings.
To avoid that I run a script that continously re-enables the mentioned settings. (mainly the headset)
It looks something like this
Code:
#!/bin/bash
while true; do
[INDENT]
amixer -c 0 csent numid=74 1
amixer -c 0 csent numid=73 1
amixer -c 0 csent numid=35 1
amixer -c 0 csent numid=52 1
amixer -c 0 csent numid=33 1
sleep 0.5
[/INDENT]
done
This probably isn't the best way of doing it (it might actually be the worst), but it works.
It might put load on the cpu or it might not, but for me, running this decreases my SoD frequency. (idek why)
Also the best way to run the script is a few seconds after the system started using the "nohup" command. That way you can run it and close the terminal and it will still be active. To kill it, search its pid using
Code:
ps -x
and kill it with
Code:
kill (insert pid here)
If you want you can also run it as a start up script.
Though if you do that, you might see that it runs but doesn't do anything.
Don't panic. Or do if you want to.
But just restart the script anyways.
OPTIONAL: (because I force you to use the stuff above)
We can play no more than ONE audio source at a time
Soloution: PulseAudio
Or so I thought... I can't seem to get it working. If someone can, please tell me how to.
So instead we have
Solution 2: Dmixer
Dmixer is what it name sounds like. It downmixes multiple sources into a single audio stream.
And it works (sometimes).
So here is how to.
Simply copy the content of the file below into a new file called .asoundrc in /home/username
Code:
pcm.!default {
type plug
slave.pcm "dmixer"
}
pcm.dmixer {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
This one is mainly pulled directly from the alsa project site and should work most of the time.
hw:0,0 might need changing on different devices. Same with card 0.
run
Code:
aplay -l
and use that to find the correct card and device.
???
The standard: I am not responsible for anything that happens to you, your phone or anything else.
Also I didn't invent any of this stuff myself. All credit goes to the original developers.
I just wrote this guide (if you can call it that)
Oh...and if you can't understand my english, well.... I'm sorry but i can't do better.
Click to expand...
Click to collapse
To use Pulseaudio, I found someone who used the Simple Protocol module to get sound and I'm now able to use MPD + MPDroid to play music from my Arch Linux chroot. Unfortunately, the playback get choppy when I open a on a single core phone with less that 1GB of RAM so if you have a better device, I'm sure the playback will be way better!
The phone I'm using is a Samsung Galaxy Ace II X (GT-S7560M) with LinuxDeploy (and CM11)
http://kaytat.com/blog/?page_id=301

TheST4RL said:
To use Pulseaudio, I found someone you used the Simple Protocol module to get sound and I'm now able to use MPD + MPDroid to play music from my Arch Linux chroot. Unfortunately, the playback get choppy when I open a on a single core phone with less that 1GB of RAM so if you have a better device, I'm sure the playback will be way better!
The phone I'm using is a Samsung Galaxy Ace II X (GT-S7560M) with LinuxDeploy (and CM11)
http://kaytat.com/blog/?page_id=301
Click to expand...
Click to collapse
I wish you would have told me this yesterday.
I spend all of yesterday doing the simple protocol player method. (It works on a quadcore 4gb phone.)
However I still prefer the alsa method on low performance phones because it has nearly no latency and is very smooth.
I will try the MPD method and see how it goes.

ruleh said:
I wish you would have told me this yesterday.
I spend all of yesterday doing the simple protocol player method. (It works on a quadcore 4gb phone.)
However I still prefer the alsa method on low performance phones because it has nearly no latency and is very smooth.
I will try the MPD method and see how it goes.
Click to expand...
Click to collapse
I tried ALSA today and it's working but the playback is super choppy and when I listen to music the audio is slower when compared to the real one. Pulseaudio may have some latency but if you're listening to music, it doesn't really matters.

Need help --> Squelch function

I found this post here: http://forum.xda-developers.com/showpost.php?p=34644359&postcount=4899
- Squelch function, mostly intended for listening to sporadic emergency service broadcasts (Japan band in Germany), but may also be useful for anyone who wants audio muted when RSSI is low.
Set Menu-> Settings-> Alternate Frequency/Squelch-> AF Mode to "Squelch". Ensure Alt Freq = 0 and set AF RSSI Threshold to desired RSSI value. Default 16 seems OK on most FM chips.
Now, the RSSI will be checked every 0.5 seconds by default. Set AF Period to desired period in milliseconds to set. If RSSI is low, audio is muted, if high, un-muted.
Click to expand...
Click to collapse
I can't find that options in my Spirit2 App. Is there a way to get this working on Spirit2 or is this just a feature of Spirit1?

kl2e said:
I found this post here: http://forum.xda-developers.com/showpost.php?p=34644359&postcount=4899
I can't find that options in my Spirit2 App. Is there a way to get this working on Spirit2 or is this just a feature of Spirit1?
Click to expand...
Click to collapse
No, it's just a Spirit1 feature.
Spirit1 pretty much has all features of Spirit2, except the new UI, BTW.

mikereidis said:
No, it's just a Spirit1 feature.
Spirit1 pretty much has all features of Spirit2, except the new UI, BTW.
Click to expand...
Click to collapse
Is Spirit1 running on a LG G3 D855 with CM 12 ?!?

kl2e said:
Is Spirit1 running on a LG G3 D855 with CM 12 ?!?
Click to expand...
Click to collapse
I'm getting this Error on Spirit UL:
Error: NO FM accessible
SU: 2 Fm: ?? Su:1 Bt:0 Ht:0
Click to expand...
Click to collapse
Which Audio Method or FM-Chip do I have to configure to get this working... ???

kl2e said:
I'm getting this Error on Spirit UL:
Which Audio Method or FM-Chip do I have to configure to get this working... ???
Click to expand...
Click to collapse
I sent you a beta of Spirit1 that is needed for Lollipop Android 5+, and instructions to enable Squelch. But it may not work on all ROMs, and on some, SELinux may need to be disabled.
LG G3 is Qualcomm I think, But I've had little feedback about it.
Default audio is "Non-Digital" (actually, it is, but can't use EQ/BT etc.) If you want full digital audio like on Spirit2, set Audio-> Method to Digital Auto or Digital QCom near end of list.
Default settings are usually fine, but FM Chip is called "QC V4L (HTC)" on Spirit1.

[PSA] You can output Hi-Res audio via internal DAC

I've found recently that in last alphas of Poweramp player you can output hi-res audio (up to 24bit/192khz supported by our chip ) directly via Snapdragon's DAC (to heaphones ). And it works! BUT!
It only works with stock Roms apparently, I've checked few Nougats, Sultan's cm13 and only stock cm13 worked....
If someone knows of any other Rom which works with this please let me know!
{
"lightbox_close": "Close",
"lightbox_next": "Next",
"lightbox_previous": "Previous",
"lightbox_error": "The requested content cannot be loaded. Please try again later.",
"lightbox_start_slideshow": "Start slideshow",
"lightbox_stop_slideshow": "Stop slideshow",
"lightbox_full_screen": "Full screen",
"lightbox_thumbnails": "Thumbnails",
"lightbox_download": "Download",
"lightbox_share": "Share",
"lightbox_zoom": "Zoom",
"lightbox_new_window": "New window",
"lightbox_toggle_sidebar": "Toggle sidebar"
}
Most of my audio is high def, plus I have pair of so called "hi-res" heaphones , one of them are cheapo Xiaomi Hybrids Pro, and this stack really does make difference. I've left Viper and Co. for good when it comes to my music.

Small update, it works wih CM13 from jgcap So that's what I'm staying on for the time being

You have hi-res headphones...
Would you consider the Sony MDR-1000X NC Headphones as "hi-res" headphones?
I have those, would I notice a difference?

It's a known Poweramp bug that should be fixed in Poweramp's next update. Currently it doesn't support hires audio on any Nougat device.

To force it to use high res on nougat, install arise, then go to /etc/init.d/ and rename arisesound_disabled_setprop to arisesound_setprop. Install version 704 alpha of Poweramp (if you haven't already), and reboot. It should now work. If that doesn't work, make sure your rom supports init.d.

r0tati0n said:
You have hi-res headphones...
Would you consider the Sony MDR-1000X NC Headphones as "hi-res" headphones?
I have those, would I notice a difference?
Click to expand...
Click to collapse
They are indeed Hi-Res certified, and have specs to follow. With decent quality recordings you might, it depends on how good is your hearing too
f41lbl0g said:
To force it to use high res on nougat, install arise, then go to /etc/init.d/ and rename arisesound_disabled_setprop to arisesound_setprop. Install version 704 alpha of Poweramp (if you haven't already), and reboot. It should now work. If that doesn't work, make sure your rom supports init.d.
Click to expand...
Click to collapse
I'm systemless ( for Android Pay ) is there systemless Arise?
Edit. Seems that, there is. Now gotta figure whatt is it about and how to install, never used it before.
Edit2. Done [just core], it sorta works now? Outputs hires to Lenovo Hi-Fi which is what you meant I guess? Won't give me peace of mind though as it outputs only in 16 bit (coverting my 24 bit audio midway).
Edit3. Sound seems actually better with that solution, it's especially noticeable with bass , it seems to truly output to DAC, latencies are even shorter than with poweramp's qualcom mode, and it seems to enable FLT_EXTENDED_DYN_RANGE in DAC, which I'm all about obviously, and that's what would result in improved bass
( I can set my bass to brain drilling now and it stays stable/without any distortions, just ...wow )

Update, probably final. Although the ARISE trick allows for forcing Hi-res option, and output, from poweramp. (resulting in pleasant sound) it doesn't use DAC directly and goes through software chain resulting in 16/48 audio in the end.
It's easy to check through adb with "adb shell dumpsys media.audio_flinger"
Native DAC output on jgcap's CM
Code:
Output thread 0xac4fe000 type 4 (OFFLOAD):
Thread name: AudioOut_17
I/O handle: 23
TID: 6203
Standby: no
Sample rate: 96000 Hz
HAL frame count: 30720
HAL format: 0x1a000004 (pcm-24bit-offload)
HAL buffer size: 30720 bytes
Channel count: 2
Channel mask: 0x00000003 (front-left, front-right)
Format: 0x1a000004 (pcm-24bit-offload)
Frame size: 1 bytes
Pending config events: none
Output device: 0x8 (WIRED_HEADPHONE)
Input device: 0 (NONE)
Audio source: 0 (default)
Normal frame count: 30720
Last write occurred (msecs): 3
Total writes: 196
Delayed writes: 0
Blocked in write: yes
Suspend count: 0
Sink buffer : 0xae0bd000
Mixer buffer: 0xae203000
Effect buffer: 0xae083000
Fast track availMask=0xfe
Standby delay ns=1000000000
AudioStreamOut: 0xae122540 flags 0x31 (DIRECT|COMPRESS_OFFLOAD|NON_BLOCKING)
Stream volumes in dB: 0:-10, 1:-27, 2:-20, 3:-17, 4:-13, 5:-20, 6:0, 7:-27, 8:-21, 9:-96, 10:-17, 11:0, 12:0
Normal mixer raw underrun counters: partial=0 empty=0
1 Tracks of which 1 are active
Name Active Client Type Fmt Chn mask Session fCount S F SRate L dB R dB Server Main buf Aux Buf Flags UndFrmCnt
none yes 6160 6 1A000004 00000003 22 61440 A 3 96000 0 0 005BA520 0xae0bd000 0x0 0x000 45792
0 Effect Chains
And on Nougat with ARISE
Code:
Output thread 0xb6179000 type 0 (MIXER):
Thread name: AudioOut_15
I/O handle: 21
TID: 671
Standby: no
Sample rate: 48000 Hz
HAL frame count: 1920
HAL format: 0x1 (pcm16)
HAL buffer size: 7680 bytes
Channel count: 2
Channel mask: 0x00000003 (front-left, front-right)
Processing format: 0x1 (pcm16)
Processing frame size: 4 bytes
Pending config events: none
Output device: 0x8 (WIRED_HEADPHONE)
Input device: 0 (NONE)
Audio source: 0 (default)
Normal frame count: 1920
Last write occurred (msecs): 16
Total writes: 30172
Delayed writes: 0
Blocked in write: yes
Suspend count: 0
Sink buffer : 0xb617b000
Mixer buffer: 0xb5311000
Effect buffer: 0xb617d000
Fast track availMask=0xfe
Standby delay ns=3000000000
AudioStreamOut: 0xb6149348 flags 0x8 (DEEP_BUFFER)
Frames written: 57930240
Suspended frames: 0
Hal stream dump:
Thread throttle time (msecs): 1228
AudioMixer tracks: 0x00000003
Master mono: off
FastMixer not initialized
Stream volumes in dB: 0:-10, 1:-31, 2:-35, 3:-28, 4:-28, 5:-35, 6:0, 7:-31, 8:-25, 9:-96, 10:-28, 11:0, 12:0
Normal mixer raw underrun counters: partial=0 empty=0
2 Tracks of which 1 are active
Name Active Client Type Fmt Chn mask Session fCount S F SRate L dB R dB Server Main buf Aux Buf Flags UndFrmCnt
1 yes 29842 3 00000005 00000003 497 7688 A 3 96000 0 0 010FE000 0xb617b000 0x0 0x001 3844
0 no 29842 3 00000001 00000003 497 7688 I 0 96000 0 0 00000000 0xb617b000 0x0 0x000 0
0 Effect Chains
In the result I'll stay with the CM13 till next poweramp update. With direct DAC output battery lasts forever, and sound quality is impossible to beat with any current software tricks.
Better bass was only due to limiter in the chain, turning it on in poweramp results in same bass improvemnt.

WilderSu said:
I've found recently that in last alphas of Poweramp player you can output hi-res audio (up to 24bit/192khz supported by our chip ) directly via Snapdragon's DAC (to heaphones ). And it works! BUT!
It only works with stock Roms apparently, I've checked few Nougats, Sultan's cm13 and only stock cm13 worked....
If someone knows of any other Rom which works with this please let me know!
Most of my audio is high def, plus I have pair of so called "hi-res" heaphones , one of them are cheapo Xiaomi Hybrids Pro, and this stack really does make difference. I've left Viper and Co. for good when it comes to my music.
Click to expand...
Click to collapse
Hi Wilder,
Can you please tell me, in the latest build (Build 798, which JUST came out I'm told), is there an option to BYPASS Snapdragon/Android audio processing if there is a separate (and superior) DAC detected (either internal or external via cable or BT)?? USB Audio Player Pro has this feature, and I'm wondering if Poweramp does as well.
This option you're speaking of appears to only be for the Snapdragon DACs.
Thank you!!

RockStar2005 said:
Hi Wilder,
Can you please tell me, in the latest build (Build 798, which JUST came out I'm told), is there an option to BYPASS Snapdragon/Android audio processing if there is a separate (and superior) DAC detected (either internal or external via cable or BT)?? USB Audio Player Pro has this feature, and I'm wondering if Poweramp does as well.
This option you're speaking of appears to only be for the Snapdragon DACs.
Thank you!!
Click to expand...
Click to collapse
Just tested it, I'm on Lineage 15.1 and not only it doesn't offer DAC but under LOS it doesn't ever offer the new Hi-Res OpenSL option. So it's a massive regression. I'm not going to test it on old CM stock, it's just not worth it. If I remember correctly latets DU offered HI-Res OpenSL [no dac though] .
Dissapointing.

WilderSu said:
Just tested it, I'm on Lineage 15.1 and not only it doesn't offer DAC but under LOS it doesn't ever offer the new Hi-Res OpenSL option. So it's a massive regression. I'm not going to test it on old CM stock, it's just not worth it. If I remember correctly latets DU offered HI-Res OpenSL [no dac though] .
Dissapointing.
Click to expand...
Click to collapse
Hey Wilder,
Thanks for starting this thread.
Which version are you testing though? The official V2 of Poweramp does NOT have any of the features I mentioned, but the last V3 "Beta/Preview" version does.
They just THIS MORNING released the Beta Build 799 version, which eliminated ALL the bugs found on previous versions from what I understood. I feel V3 is coming VERY VERY soon based on these recent updates which have come faster than usual. (V2 came out April 2013 FYI lol). I would recommend going to Play Store, signing up for Beta on the Poweramp trial page in Play Store, and then when the update appears update it, and enjoy.
IF you do this, please try both the OpenSL ES Output and HI-Res Output options and see if you can determine which one actually DOES the bypassing in question (if either do). I think we both would like to know. lol
If not, that's fine too.
Also, this article (below) from 2 months ago that I found earlier today claims ALL the apps listed in it offer bypassing of Android audio, at least that's how I'm interpreting. It mentions an earlier build of the V3 Beta Poweramp, so the 799 will definitely be a stable version of it to try out. Check it out and let me know what you think too.
https://www.google.com/search?q=The...=chrome..69i57j69i65&sourceid=chrome&ie=UTF-8 (CHOOSE THE FIRST ONE TO SEE ARTICLE!)
Thanks either way!!

RockStar2005 said:
Hey Wilder,
Thanks for starting this thread.
Which version are you testing though? The official V2 of Poweramp does NOT have any of the features I mentioned, but the last V3 "Beta/Preview" version does.
They just THIS MORNING released the Beta Build 799 version, which eliminated ALL the bugs found on previous versions from what I understood. I feel V3 is coming VERY VERY soon based on these recent updates which have come faster than usual. (V2 came out April 2013 FYI lol). I would recommend going to Play Store, signing up for Beta on the Poweramp trial page in Play Store, and then when the update appears update it, and enjoy.
IF you do this, please try both the OpenSL ES Output and HI-Res Output options and see if you can determine which one actually DOES the bypassing in question (if either do). I think we both would like to know. lol
If not, that's fine too.
Also, this article (below) from 2 months ago that I found earlier today claims ALL the apps listed in it offer bypassing of Android audio, at least that's how I'm interpreting. It mentions an earlier build of the V3 Beta Poweramp, so the 799 will definitely be a stable version of it to try out. Check it out and let me know what you think too.
https://www.google.com/search?q=The...=chrome..69i57j69i65&sourceid=chrome&ie=UTF-8 (CHOOSE THE FIRST ONE TO SEE ARTICLE!)
Thanks either way!!
Click to expand...
Click to collapse
I'm on latest v3 and it doesn't work, checked 12.5 DU and RR 6 ROMs and it doesn't work there either, seems the dev just checks it against latest official stock which was 5.1.1 CM which sucks majorly.

WilderSu said:
I'm on latest v3 and it doesn't work, checked 12.5 DU and RR 6 ROMs and it doesn't work there either, seems the dev just checks it against latest official stock which was 5.1.1 CM which sucks majorly.
Click to expand...
Click to collapse
Ok I didn't understand like 90% of that. LOL But thanks anyway.
So you think it'll work on the official V3 then??
Uggh this means if I notice a difference (with bypass on vs. off) I'd have to use UAPP. But I like Poweramp more, damm*t!!!!!! LOL

RockStar2005 said:
Ok I didn't understand like 90% of that. LOL But thanks anyway.
So you think it'll work on the official V3 then??
Uggh this means if I notice a difference (with bypass on vs. off) I'd have to use UAPP. But I like Poweramp more, damm*t!!!!!! LOL
Click to expand...
Click to collapse
Don't think it will, unless enough users will spam their forum about this phone. It's just too old, no one cares anymore.

WilderSu said:
Don't think it will, unless enough users will spam their forum about this phone. It's just too old, no one cares anymore.
Click to expand...
Click to collapse
Hmm ok.
What about that article? It claims it can do it though?

Wasn't there a DAC enabler for Snapdragon Chipsets in Magisk Manager?
There is aptX HD available through Bluetooth on Oreo ROMS and newer.
I don't know if it works on Nougat or below though as I'm not an audiophile.

Deleted