Eliminating the loud hum from powered speakers connected to my laptop

My external powered speakers emit a loud, annoying hum if I plug them into the headphones socket of my Clevo W230SS laptop. The hum stops if I unplug the laptop’s mains charger, so the problem is due to an earth loop (a.k.a. ‘ground loop’). The charger’s recessed socket for connecting its mains lead has three prongs (IEC 60320 C6), i.e. the charger is not double-insulated. The recessed socket on a double-insulated mains charger would only have two prongs (IEC 60320 C8), and the label on a double-insulated charger in the UK usually has an icon consisting of two concentric squares. I suspect the audio circuitry in the Clevo W230SS is not very well designed, as there is virtually no hum from the same powered speakers if I plug them into the headphones socket of my Compal NBLB2 laptop while its mains charger, which has the same IEC 60320 C5/C6 coupler design as the Clevo laptop’s charger, is connected.

Now, I could have purchased a ground loop isolator, a passive device which is basically a couple of transformers (one per stereo channel), to interpose between the headphones socket on my laptop and the external powered speakers.

The YouTube video Solving Laptop Noise and Ground Loop Audio Interference contains a demonstration of what happens when you use a ground loop isolator on the headphones output of a laptop, and what happens when you use a double-insulated mains charger (two-prong coupler, rather than three-prong).

However, rather than buy a ground loop isolator I decided to try an active device, and purchased the XQ-10 portable headphones amplifier manufactured by Chinese company xDuoo. The XQ-10 is compact, fitting in the palm of my hand and weighing only 30 grammes. It has an aluminium-alloy case, and looks and feels well-made. Two cables are supplied: a USB cable to connect the XQ-10 to a phone charger or laptop in order to charge the LiPo battery in the XQ-10 (about 2 hours to charge, and about 20 hours of use on battery power), and a short 3.5 mm jack cable (male jack at both ends) to connect the XQ-10 to your laptop or phone.

xDuoo XQ-10

If the XQ-10’s USB charging cable is not connected (see the photograph above) then the loud hum from my external powered speakers still occurs. But with both cables connected (see the photograph below), the XQ-10 does the job perfectly: there is no hum or hiss, even when I turn the speakers’ volume knob to maximum.

xDuoo XQ-10

Although I do not need to use a headphones amplifier with the laptop when I use headphones or earphones — the volume is fine and there is no hum from headphones/earphones — I should mention that audio quality from my headphones and earphones connected to the XQ-10 is excellent (both with and without the USB charging cable connected, which is not surprising given that no earth loop exists when headphones/earphones are connected).

By the way, shop around if you do decide to buy the XQ-10, because its price on-line varies a lot. I paid GBP 17.95 for mine, and that included p&p.

No sound from headphones after resume from suspension / No sound from headphones after re-plug

It is not difficult to find posts on the Web regarding certain models of laptop that no longer produce sound from headphones after resuming from suspension, or no longer produce sound from their speakers or from headphones if you unplug and reconnect the headphones. My Clevo W230SS laptop suffered from these problems and more: sometimes the external microphone socket would no longer work either. I had to reboot the laptop in order to get audio working properly again.

The cause of these problems varies according to the specific hardware and software, and here I will describe a couple of fixes I implemented in Gentoo Linux for my Clevo W230SS laptop. Bear in mind that what works for one model of laptop may not necessarily work for a different model even if the symptoms are the same.

PROBLEM 1: No sound from headphones after resume from suspension

After my laptop resumed from suspension, headphones would no longer work until I rebooted the laptop. Sometimes an external microphone would also stop working until I rebooted. In 2014, Ubuntu user Kiril filed a bug report regarding this problem with the Clevo W230SS: [W230SS, VIA VT1802, Green Headphone Out, Front] No sound after suspend/resume. Actually, his original title for the bug report was: ‘[W230SS, VIA VT1802, Green Headphone Out, Front] No sound after fresh boot’. I didn’t have that problem: the headphone socket of my Clevo W230SS did produce sound after a ‘fresh boot’. Regarding Kiril‘s initial problem, ALSA developer Raymond Yau made several comments, including the following:

driver should not use same audio output for device 0 and device 2

independent headphone should be disabled on notebook by default

for desktop line out and headphone connected to different audio output nodes 0x03 and 0x04

this allow you to play different audio to line out and headphone

but this feature usually should be disabled for notebook

you need to file an upstream bug report to fix this bug in the indep_hp_possible function which should return false when there is no internal mic since some notebook have line out, headphone and Mic jack to support 5.1 or only enabled when headphone at extra front and line out at ext rear

the workaround is use hint to disable the indep_hp=0

https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documentation/sound/alsa/HD-Audio.txt

Now, Kiril was using Ubuntu 14.04 and ALSA 1.0.27, whereas I’m using Gentoo and ALSA 1.0.29. Furthermore, apart from the two installations using different kernel versions, in Gentoo you configure and build the kernel yourself. So there is quite some difference between the two installations, which might explain why I do not have his original problem of no sound from headphones after a fresh boot. Where we did coincide, though, was that there was no sound from headphones following resumption from suspension.

First attempt at fixing the problem

Even though Independent HP does not stop headphones working after I boot my laptop, I decided to try to remove Independent HP anyway, to see if it would fix the suspend/resume problem with headphones in my case.

The Clevo W230SS has two sound cards:

root # lspci | grep Audio
00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06)
00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 05)

The HDMI audio card is Card 0, and the analogue HD Audio card is Card 1. ALSAMixer shows an Independent HP (headphone) channel for Card 1:

user $ alsamixer -c 1

The kernel documentation for the HD Audio driver explains how to fix the problem using what is called a driver ‘hint’. There is a link to the documentation in the above-mentioned bug report, and you can also find the documentation in the file /usr/src/<kernel_release>/Documentation/sound/alsa/HD-Audio.txt on your laptop if you have installed the kernel source code. As explained in the documentation, you can remove Independent HP either via the command line:

root # echo "indep_hp = no" > /sys/class/sound/hwC1D0/hints
root # echo 1 > /sys/class/sound/hwC1D0/reconfig

or by using so-called ‘Early Patching’:

Early Patching
~~~~~~~~~~~~~~
When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a
firmware file for modifying the HD-audio setup before initializing the
codec.  This can work basically like the reconfiguration via sysfs in
the above, but it does it before the first codec configuration.

Note that the term ‘patching’ here has nothing to do with patching the driver’s source code; it refers to patching the ALSA driver’s configuration of the VIA chip’s CODEC on the Intel sound card.

The format of this particular patch file containing a ‘hint’ would be as follows:

[codec]
<vendor_id> <subsystem_id> <address_of_the_CODEC>

[hint]
indep_hp = no

The values for vendor_id and subsystem_id can be found as follows:

root # cat /sys/class/sound/hwC1D0/vendor_id
0x11068446
root # cat /sys/class/sound/hwC1D0/subsystem_id
0x15582300

(As the driver is used with Card 1, remember to look in directory hwC1D0 rather than hwC0D0.)

The required value for address_of_the_CODEC is zero in this particular case. Therefore the file /lib/firmware/clevo-hda-patch (you can choose any file name you want) should have the following contents:

[codec]
0x11068446 0x15582300 0

[hint]
indep_hp = no

If you built the HA Audio driver as a module, you would need to add the following line to the file /etc/modprobe.d/alsa.conf in order to apply the patch:

options snd-hda-intel patch=,clevo-hda-patch

Notice the comma after the equals sign. This is required because the patch applies to the second card (Card 1) rather than to the first card (Card 0).

However, I had built the HD Audio driver into the kernel rather than as a module:

root # grep HDA /usr/src/linux/.config | grep -v "is not set"
CONFIG_SND_HDA=y
CONFIG_SND_HDA_INTEL=y
CONFIG_SND_HDA_PREALLOC_SIZE=2048
CONFIG_SND_HDA_HWDEP=y
CONFIG_SND_HDA_RECONFIG=y
CONFIG_SND_HDA_INPUT_BEEP=y
CONFIG_SND_HDA_INPUT_BEEP_MODE=1
CONFIG_SND_HDA_INPUT_JACK=y
CONFIG_SND_HDA_PATCH_LOADER=y
CONFIG_SND_HDA_CODEC_VIA=y
CONFIG_SND_HDA_CODEC_HDMI=y
CONFIG_SND_HDA_I915=y
CONFIG_SND_HDA_GENERIC=y
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0

Therefore adding the option to /etc/modprobe.d/alsa.conf would have no effect, as the HD Audio driver is not a module. In this case I could have appended the following parameter to the kernel boot line in the file /boot/grub/grub.cfg instead:

snd-hda-intel.patch=,clevo-hda-patch

The above kernel boot parameter could be appended to the kernel boot line either directly by editing the file grub.cfg, or indirectly by adding it to the list of boot parameters in the variable GRUB_CMDLINE_LINUX_DEFAULT="" in the file /etc/default/grub and then using the command ‘grub2-mkconfig -o /boot/grub/grub.cfg‘ as root user to regenerate the grub.cfg file. Remember to mount /boot first if it is on a different partition.

Now, I tried applying the patch using the appropriate method in each case: HD Audio driver built into the kernel, and HD Audio driver built as a module (it didn’t take me long to modify the kernel configuration and rebuild the kernel). In both cases the following messages were included in the output of the dmesg command:

[ 0.430218] snd_hda_intel 0000:00:1b.0: Applying patch firmware 'clevo-hda-patch'
[ 0.430356] snd_hda_intel 0000:00:1b.0: Direct firmware load for clevo-hda-patch failed with error -2
[ 0.430359] snd_hda_intel 0000:00:1b.0: Cannot load firmware, aborting

I’m not sure if the reason for the failure to load the patch is the same in both cases, but certainly the reason in the case of the module is that PulseAudio is already running and using the driver by the time the OS attempts to apply the patch. In the case of the kernel boot parameter, my guess is that the patch would need to be included in an initramfs in order to be able to apply it before PulseAudio starts.

The same situation occurs if you try to apply the ‘hint’ manually from the command line after start-up:

root # echo "indep_hp = no" > /sys/class/sound/hwC1D0/hints
root # echo 1 > /sys/class/sound/hwC1D0/reconfig
bash: echo: write error: Device or resource busy

The above error message occurs because PulseAudio is running and using the driver. To apply the hint and refresh the driver in this case, the solution is to stop PulseAudio beforehand:

user $ echo "autospawn = no" >> ~/.config/pulse/client.conf
user $ pulseaudio --kill
user $ su
root # echo "indep_hp = no" > /sys/class/sound/hwC1D0/hints
root # echo 1 > /sys/class/sound/hwC1D0/reconfig
root # exit
user $ pulseaudio --start

Now, having to do the above manually every time you boot your machine is impractical. To automate the procedure I did the following…

Make sure the automatic (re)spawning of PulseAudio is disabled in the file ~/.config/pulse/client.conf (you can create the file if it does not exist):

autospawn = no

Create a Bash script in the directory /etc/local.d/ for the OS to launch automatically at boot, and in that script issue the above two commands first and then start pulseaudio. I named the script ‘99-clevo-hda-fix.start‘ and made its contents the following:

#!/bin/bash
# Fix for Intel HDA problem with Clevo W230SS
# See https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1313904
# N.B. Assumes your ~/.config/pulse/client.conf contains
# 'autospawn = no' so that PulseAudio is not launched automatically.
#
# Specify the hint
echo "indep_hp = no" > /sys/class/sound/hwC1D0/hints
#
# Reinitialise the HD Audio driver so it parses the CODEC tree again
echo 1 > /sys/class/sound/hwC1D0/reconfig
#
# Start PulseAudio for my user account
su -c "pulseaudio --start" -s /bin/sh fitzcarraldo

Don’t forget to make the script executable:

root # chmod +x /etc/local.d/99-clevo-hda-fix.start

This method of applying the hint to the HD Audio driver should work irrespectively of whether the driver is a module or built-in. It’s a bit more of a hack compared to the Early Patching approach, but it does the job in my case. After logging in to the Desktop Environment, you can check if the hint has been applied by looking at the contents of /sys/class/sound/hwC1D0/hints:

root # cat /sys/class/sound/hwC1D0/hints
indep_hp = no

However, this alone does not mean the CODEC was reconfigured after the hint was applied, so you need to check if Independent HP still exists. The ALSAMixer output on my laptop now looked like the following after I implemented the above method (notice that Independent HP no longer exists):

user $ alsamixer -c 1

So, what was the result? Well, audio continued to work after I removed Independent HP, but there was still no sound from headphones after resuming from suspension. And neither was there for Kiril, so he changed the title of his bug report to that effect. Fortunately, another user, unrud, commented later in the bug report that he had written init-headphone, a Python script to fix the problem. So I decided to hack his Ubuntu package to get init-headphone working in my Gentoo Linux installation. Here is how I did it…

Second (successful!) attempt at fixing the problem

1. Download init-headphone-ubuntu-0.11.zip from https://github.com/Unrud/init-headphone-ubuntu/releases

2. Extract the contents to the directory ~/init-headphone-ubuntu-0.11/

3. Copy to pm-utils’ hook directory the script that launches init-headphone upon resuming or thawing:

root # cp /home/fitzcarraldo/init-headphone-ubuntu-0.11/etc/linux-pm-utils/init-headphone /etc/pm/sleep.d/03-init-headphone # (use whatever number you want)

4. Copy the init-headphone script itself to the system binaries directory:

root # cp /home/fitzcarraldo/init-headphone-ubuntu-0.11/src/init-headphone /usr/local/sbin/

5. The init-headphone script requires the i2c_dev and i2c_i801 modules:

REQUIRED_MODULES = ["i2c_dev", "i2c_i801"]

However, I prefer to build them into the kernel rather than as modules, so I checked to make sure they are already built into the kernel:

root # grep CONFIG_I2C /usr/src/linux/.config | grep -v "is not set"
CONFIG_I2C=y
CONFIG_I2C_BOARDINFO=y
CONFIG_I2C_COMPAT=y
CONFIG_I2C_CHARDEV=y
CONFIG_I2C_MUX=y
CONFIG_I2C_HELPER_AUTO=y
CONFIG_I2C_ALGOBIT=y
CONFIG_I2C_I801=y

Then I commented out the lines in /usr/local/sbin/init-headphone that check if the two modules are loaded:

root # diff /home/fitzcarraldo/init-headphone-ubuntu-0.11/src/init-headphone /usr/local/sbin/init-headphone
174,181c174,181
< def check_modules():
< try:
< for module in REQUIRED_MODULES:
< logging.info("Trying to add module to the kernel: %s", module)
< if subprocess.call(["modprobe", "--quiet", module]) != 0:
< logging.warning("Module is not loaded: %s", module)
< except OSError:
#def check_modules():
> # try:
> # for module in REQUIRED_MODULES:
> # logging.info("Trying to add module to the kernel: %s", module)
> # if subprocess.call(["modprobe", "--quiet", module]) != 0:
> # logging.warning("Module is not loaded: %s", module)
> # except OSError:
> # logging.warning("modprobe not found")
206c206
# check_modules()

6. I created a script /etc/local.d/99-clevo-hda-fix.start to launch init-headphone automatically at boot:

#!/bin/bash
# Fix for Intel HDA problem with Clevo W230ss
# See https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1313904
exec /usr/local/sbin/init-headphone

root # chmod +x /etc/local.d/99-clevo-hda-fix.start

7. Add the kernel boot parameter ‘acpi_enforce_resources=lax‘ to the end of the kernel boot line(s) in /boot/grub/grub.cfg (don’t forget to mount /boot first if it is on another partition). You can either edit /boot/grub/grub.cfg directly, or indirectly by adding the parameter to the list of existing parameters in GRUB_CMDLINE_LINUX_DEFAULT (if any) as shown below and issuing the command ‘grub2-mkconfig -o /boot/grub/grub.cfg‘ as root user (again, don’t forget to mount /boot first if it is on another partition):

GRUB_CMDLINE_LINUX_DEFAULT="acpi_enforce_resources=lax"

8. Create a script file /etc/local.d/99-clevo-hda-fix.start to launch the init-headphone automatically at boot:

#!/bin/bash
# Fix for Intel HDA problem with Clevo W230SS
# See https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1313904
exec /usr/local/sbin/init-headphone

9. As my user account does not have a path configured to the system binaries directory, I created a symlink to it from the directory /usr/local/bin/:

root # ln -s /usr/local/sbin/init-headphone /usr/local/bin/init-headphone

10. As I no longer needed to stop PulseAudio running at boot, I deleted the file /home/fitzcarraldo/.config/pulse/client.conf I had created earlier (or you could just change ‘autospawn = no‘ to ‘autospawn = yes‘).

You can also use init-headphone from the command line:

user $ sudo init-headphone --help

e.g.

user $ sudo init-headphone unmute

This looks like it has finally solved the problem; now headphones still work after my laptop resumes from suspension. I still don’t need to pass the ‘hint’ to the HD Audio driver, so Independent HP continues to appear in ALSAMixer and apparently does not cause any problems.

A big ‘Thank you’ from me to Unrud for creating init-headphone.

PROBLEM 2: No sound from headphones after re-plug

The other problem I experienced with the Clevo W230SS was that, if I unplugged working headphones, audio switched to the laptop’s speakers as expected, but, if I then plugged-in the headphones again, no more sound came from the headphones. If I again unplugged the headphones, sound would again come from the speakers. If I did all this whilst ALSAMixer was running, then:

  1. if I unplugged the headphones, as expected the ALSAMixer volume level indicator for the speaker would rise from zero and the volume level indicator for the headphones would drop to zero;
  2. if I plugged in the headphones, as expected the ALSAMixer volume level indicator for the speaker would drop to zero and the volume level indicator for the headphones would rise from zero.

Now, it is possible that this problem was due to the same thing that caused the loss of audio to headphones when the laptop resumed from suspension. Anyway, before I came across init-headphone I found the following in the Arch Linux Wiki article on PulseAudio:

Switch on connect

This is a module used to switch the output sound to the newly connected device. For example, if you plug in a USB headset, the output will be switched to that. If you unplug it, the output will be set back to the last device. This used to be quite buggy but got a lot of attention in PulseAudio 8.0 and should work quite well now.

If you just want to test the module then you can load it at runtime by calling:

root # pactl load-module module-switch-on-connect

If you want to make the change persistent you will have to add it to your local pulseaudio settings or to /etc/pulse/default.pa (system wide effect). In either case, add this line:

load-module module-switch-on-connect

So, as the file /etc/pulse/default.pa in my installation did not have that line, I added it:

# Added by fitzcarraldo
# https://wiki.archlinux.org/index.php/PulseAudio
# The headphone socket no longer worked if I
# removed and re-inserted the jack plug.
load-module module-switch-on-connect

This seemed to help, but I am not certain module-switch-on-connect is really having an effect when I plug and unplug headphones, and I have not bothered to disable it to see what happens now that init-headphone is in use (I’m just happy that audio is all working now, whatever the reason!). ALSA’s Auto-Mute Mode* appears to perform the same role as module-switch-on-connect: When I unplug the headphones with Auto-Mute Mode enabled in ALSAMixer, there is a noticeable delay before sound starts to come from the laptop’s speakers, whereas there is no such delay when I unplug the headphones with Auto-Mute Mode disabled via ALSAMixer, so presumably module-switch-on-connect is doing its job.

* The documentation file /usr/src/<kernel_release>/Documentation/sound/alsa/HD-Audio-Controls.txt explains what Auto-Mute Mode does.

Anyhow, one or both of the two software modifications (init-headphone and module-switch-on-connect) seem to have cured the problem of no sound from headphones after they are disconnected then reconnected to the laptop.

Change the mp3 bitrate for ripping in K3b

I’ve been using K3b successfully (it’s currently at version 2.0.3-r1 in Gentoo Linux) to rip Audio CDs to mp3 files, but despite changing the bitrate to 192 kbps in the ‘K3b Lame Mp3 Encoder’ plugin settings, K3b was still ripping mp3 files at 128 kbps. I found out that I needed to make another change too.

The default bitrate for ripped mp3 tracks is 128 kbps in K3b. To use a different bitrate, I needed to do the following:

1. ‘Settings’ > ‘Configure K3b…’

2. Click on ‘Plugins’

3. Click on the spanner next to ‘K3b Lame Mp3 Encoder’, and change the bitrate on the Settings tab to 192 (or whatever you want). However, this alone does not have any effect, so also click on the spanner next to ‘K3b External Audio Encoder’, click on ‘Mp3 (Lame)’ in the list of ‘Configured Encoders’, click on ‘Edit’ and insert ‘-b 192‘ (or whatever bitrate you want) in the list of lame options, like so:

lame -r --bitwidth 16 --little-endian -b 192 -s 44.1 -h --tt %t --ta %a --tl %m --ty %y --tc %c --tn %n - %f

Here’s what happens before I added the ‘-b 192‘ to the settings for ‘Mp3 (Lame)’ in ‘K3b External Audio Encoder’:

$ file 01\ -\ Meu\ Bem\ Querer.mp3
01 - Meu Bem Querer.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, JntStereo

And after adding the ‘-b 192‘ to the settings for ‘Mp3 (Lame)’ in ‘K3b External Audio Encoder’:

$ file 01\ -\ Meu\ Bem\ Querer.mp3
01 - Meu Bem Querer.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, JntStereo

Notice the bit rate of the file has changed.

Downloading YouTube video or only audio

One of my favourite Linux applications is the command-line utility youtube-dl, which I use quite often to download YouTube videos or extract only the audio, either for leisure or work purposes.

Downloading a video

  1. Copy the URL of the YouTube video from the browser’s address bar. I’ll use the video ‘Como fazer um bom churrasco de Picanha‘ as an example.

  2. Ascertain the types and resolutions available for that particular video:

    $ youtube-dl -F https://www.youtube.com/watch?v=8BuQCD8eh9k
    8BuQCD8eh9k: Downloading webpage
    8BuQCD8eh9k: Extracting video information
    8BuQCD8eh9k: Downloading DASH manifest
    [info] Available formats for 8BuQCD8eh9k:
    format code extension resolution  note
    171         webm      audio only  DASH audio  118k , audio@128k (44100Hz), 5.05MiB
    140         m4a       audio only  DASH audio  129k , m4a_dash container, aac @128k (44100Hz), 5.99MiB
    141         m4a       audio only  DASH audio  256k , m4a_dash container, aac @256k (44100Hz), 11.90MiB
    160         mp4       256x144     DASH video  111k , 15fps, video only, 5.03MiB
    242         webm      426x240     DASH video  190k , 1fps, video only, 7.19MiB
    133         mp4       426x240     DASH video  248k , 30fps, video only, 11.33MiB
    243         webm      640x360     DASH video  378k , 1fps, video only, 13.83MiB
    134         mp4       640x360     DASH video  465k , 30fps, video only, 12.98MiB
    244         webm      854x480     DASH video  744k , 1fps, video only, 27.11MiB
    135         mp4       854x480     DASH video  873k , 30fps, video only, 25.98MiB
    247         webm      1280x720    DASH video 1378k , 1fps, video only, 49.45MiB
    136         mp4       1280x720    DASH video 1837k , 30fps, video only, 58.40MiB
    17          3gp       176x144
    36          3gp       320x240
    5           flv       400x240
    43          webm      640x360
    18          mp4       640x360
    22          mp4       1280x720 (best)

  3. Chose which video resolution you would like to download. For example I’ll download the highest resolution available for this particular video (Format 22, which is a 1280×720 MPEG-4 file):

    $ youtube-dl -f 22 -o Como_fazer_um_bom_churrasco_de_Picanha.mp4 https://www.youtube.com/watch?v=8BuQCD8eh9k
    8BuQCD8eh9k: Downloading webpage
    8BuQCD8eh9k: Extracting video information
    8BuQCD8eh9k: Downloading DASH manifest
    [download] Destination: Como_fazer_um_bom_churrasco_de_Picanha.mp4
    [download] 100% of 83.77MiB in 07:09

Extracting audio from YouTube videos

Let’s say you just want an audio file of the video’s sound track. There are several options, so I’ll just show a few simple examples here:

  1. MP3 file at default quality:

    $ youtube-dl --extract-audio --audio-format mp3 -t https://www.youtube.com/watch?v=8BuQCD8eh9k
    8BuQCD8eh9k: Downloading webpage
    8BuQCD8eh9k: Extracting video information
    8BuQCD8eh9k: Downloading DASH manifest
    [download] Destination: Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a
    [download] 100% of 11.90MiB in 00:56
    [ffmpeg] Correcting container in "Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a"
    [ffmpeg] Destination: Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.mp3
    Deleting original file Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a (pass -k to keep)

    You can see from the console output above that youtube-dl downloads an M4A file and then uses ffmpeg to convert it to the audio file type you specified. You can check the file’s properties:

    $ file Como\ fazer\ um\ bom\ churrasco\ de\ Picanha-8BuQCD8eh9k.mp3
    Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 64 kbps, 44.1 kHz, Stereo

  2. MP3 file of 128 kbps:

    $ youtube-dl --extract-audio --audio-format mp3 --audio-quality 128K -t https://www.youtube.com/watch?v=8BuQCD8eh9k
    8BuQCD8eh9k: Downloading webpage
    8BuQCD8eh9k: Extracting video information
    8BuQCD8eh9k: Downloading DASH manifest
    [download] Destination: Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a
    [download] 100% of 11.90MiB in 01:23
    [ffmpeg] Correcting container in "Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a"
    [ffmpeg] Destination: Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.mp3
    Deleting original file Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a (pass -k to keep)
    $ file Como\ fazer\ um\ bom\ churrasco\ de\ Picanha-8BuQCD8eh9k.mp3
    Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, Stereo

  3. M4A file at best quality available for that particular video:

    $ youtube-dl --extract-audio --audio-format best -t https://www.youtube.com/watch?v=8BuQCD8eh9k
    8BuQCD8eh9k: Downloading webpage
    8BuQCD8eh9k: Extracting video information
    8BuQCD8eh9k: Downloading DASH manifest
    [download] Destination: Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a
    [download] 100% of 11.90MiB in 01:07
    [ffmpeg] Correcting container in "Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a"
    Post-process file Como fazer um bom churrasco de Picanha-8BuQCD8eh9k.m4a exists, skipping

To see the many options available in youtube-dl:

$ man youtube-dl

or:

$ youtube-dl --help

Have fun!

Intel HD Audio clicks when a laptop is running on battery power

Yet another audio problem in Linux on my laptop. There were clicks in the audio output from headphones and the built-in speakers while running the laptop on battery power and listening to music. Details of my laptop’s two sound cards are given below:

# lspci | grep -i audio
00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05)
01:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Redwood HDMI Audio [Radeon HD 5000 Series]

# hwinfo --sound
17: PCI 1b.0: 0403 Audio device
[Created at pci.328]
Unique ID: u1Nb.sEm8vDosIQF
SysFS ID: /devices/pci0000:00/0000:00:1b.0
SysFS BusID: 0000:00:1b.0
Hardware Class: sound
Model: "Intel 5 Series/3400 Series Chipset High Definition Audio"
Vendor: pci 0x8086 "Intel Corporation"
Device: pci 0x3b56 "5 Series/3400 Series Chipset High Definition Audio"
SubVendor: pci 0x14c0 "COMPAL Electronics Inc"
SubDevice: pci 0x0043
Revision: 0x05
Driver: "snd_hda_intel"
Driver Modules: "snd_hda_intel"
Memory Range: 0xd8100000-0xd8103fff (rw,non-prefetchable)
IRQ: 31 (17929 events)
Module Alias: "pci:v00008086d00003B56sv000014C0sd00000043bc04sc03i00"
Driver Info #0:
Driver Status: snd_hda_intel is active
Driver Activation Cmd: "modprobe snd_hda_intel"
Config Status: cfg=new, avail=yes, need=no, active=unknown

28: PCI 100.1: 0403 Audio device
[Created at pci.328]
Unique ID: NXNs.nzRufOXHPI0
Parent ID: 3hqH.Db2_i3v_bw2
SysFS ID: /devices/pci0000:00/0000:00:03.0/0000:01:00.1
SysFS BusID: 0000:01:00.1
Hardware Class: sound
Model: "ATI Redwood HDMI Audio [Radeon HD 5000 Series]"
Vendor: pci 0x1002 "ATI Technologies Inc"
Device: pci 0xaa60 "Redwood HDMI Audio [Radeon HD 5000 Series]"
SubVendor: pci 0x0000
SubDevice: pci 0xaa60
Driver: "snd_hda_intel"
Driver Modules: "snd_hda_intel"
Memory Range: 0xd8020000-0xd8023fff (rw,non-prefetchable)
IRQ: 32 (87 events)
Module Alias: "pci:v00001002d0000AA60sv00000000sd0000AA60bc04sc03i00"
Driver Info #0:
Driver Status: snd_hda_intel is active
Driver Activation Cmd: "modprobe snd_hda_intel"
Config Status: cfg=new, avail=yes, need=no, active=unknown
Attached to: #9 (PCI bridge)

The audio codecs in use are as follows:

# cat /proc/asound/card0/codec* | grep Codec
Codec: Realtek ALC272
# cat /proc/asound/card1/codec* | grep Codec
Codec: ATI R6xx HDMI

The kernel modules loaded are as follows:

# lsmod | grep codec
snd_hda_codec_realtek 50367 1
snd_hda_codec_hdmi 32069 1
snd_hda_codec_generic 40974 1 snd_hda_codec_realtek
snd_hda_codec 76106 5 snd_hda_codec_realtek,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_hda_int
el,snd_hda_controller
snd_hwdep 5373 1 snd_hda_codec
snd_pcm 67896 4 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_i
ntel,snd_hda_controller
snd 49659 16 snd_hda_codec_realtek,snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_pcm,snd_h
da_codec_generic,snd_hda_codec,snd_hda_intel

The Linux kernel has a power-saving feature for Intel HD Audio cards. Each time the audio card goes into the power-saving state, or wakes up from the power-saving state when playing audio, it may produce a clicking noise. Kernel parameter CONFIG_SND_HDA_POWER_SAVE_DEFAULT is used to specify the default time-out in seconds, and setting it to zero disables power-saving mode. Similarly, kernel parameter CONFIG_SND_AC97_POWER_SAVE_DEFAULT is used to specify the default time-out in seconds for AC97 power-saving mode, and setting it to zero disables AC97 power-saving mode. As you can see below, I have both of those parameters set to zero in the kernel:

# grep POWER_SAVE /usr/src/linux/.config
CONFIG_SND_AC97_POWER_SAVE=y
CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0

Yet I still heard clicks when the laptop was running under battery power. To disable the HD Audio power-save mode for good, I added ‘snd_hda_intel.power_save=0‘ to the end of the kernel boot line in /etc/default/grub and regenerated the grub.cfg file:

# grep GRUB_CMDLINE_LINUX /etc/default/grub
GRUB_CMDLINE_LINUX="BOOT_IMAGE=/kernel-genkernel-x86_64-3.17.1-gentoo-r1 root=/dev/ram0 ramdisk=8192 real_root=/dev/sda6 init=/linuxrc splash=silent,theme:Emergance console=tty1 quiet resume=swap:/dev/sda5 real_resume=/dev/sda5 intel_iommu=off net.ifnames=0 snd_hda_intel.power_save=0"

# mount /dev/sda3 /boot # My boot directory is on its own partition.
# grub2-mkconfig -o /boot/grub/grub.cfg

After rebooting, there are no more clicks when listening to audio under battery power. Problem solved.

Audio in Linux becomes annoying again (continued)

Well, it turned out that the problem playing some system sounds in Thunderbird, described in my previous post, was due to PulseAudio.

Despite sounding fine when played by SMPlayer, the audio clips that sounded distorted/scratchy and too loud when played by Thunderbird also sounded that way when played by VLC. Then I discovered several other .wav files on various Web sites that sounded distorted when played by the browser’s Windows Media Player plug-in (Gecko Media Player). So the problem clearly was not caused by Thunderbird itself. I began to wonder if PulseAudio was the cause. So I adjusted PulseAudio’s sampling frequency, number of fragments and fragment size, and all the clips that previously sounded distorted and too loud now play fine. Here is what I did to fix the problem…

1. Check what PulseAudio is configured to use:

$ pulseaudio --dump-conf
### Read from configuration file: /etc/pulse/daemon.conf ###
daemonize = no
fail = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 5
allow-module-loading = yes
allow-exit = yes
use-pid-file = yes
system-instance = no
local-server-type = user
cpu-limit = no
enable-shm = yes
flat-volumes = no
lock-memory = no
exit-idle-time = 20
scache-idle-time = 20
dl-search-path = /usr/lib64/pulse-5.0/modules
default-script-file = /etc/pulse/default.pa
load-default-script-file = yes
log-target =
log-level = notice
resample-method = auto
enable-remixing = yes
enable-lfe-remixing = no
default-sample-format = s16le
default-sample-rate = 44100
alternate-sample-rate = 48000
default-sample-channels = 2
default-channel-map = front-left,front-right
default-fragments = 4
default-fragment-size-msec = 25
enable-deferred-volume = yes
deferred-volume-safety-margin-usec = 8000
deferred-volume-extra-delay-usec = 0
shm-size-bytes = 0
log-meta = no
log-time = no
log-backtrace = 0
rlimit-fsize = -1
rlimit-data = -1
rlimit-stack = -1
rlimit-core = -1
rlimit-rss = -1
rlimit-as = -1
rlimit-nproc = -1
rlimit-nofile = 256
rlimit-memlock = -1
rlimit-locks = -1
rlimit-sigpending = -1
rlimit-msgqueue = -1
rlimit-nice = 31
rlimit-rtprio = 9
rlimit-rttime = 1000000
$

2. Check PulseAudio’s output sample rate:

$ pacmd list-sinks | grep sample
sample spec: s16le 2ch 44100Hz
$

So the sample rate is 16 bits @ 44100 Hz and there are 2 output channels (Stereo). My laptop does indeed have two built-in stereo speakers. ‘s16le‘ means ‘signed 16-bit little-endian’.

3. Check PulseAudio’s input sample rate:

$ pacmd list-sources | grep sample
sample spec: s16le 2ch 44100Hz
sample spec: s16le 2ch 44100Hz
$

So the sample rate is 16 bits @ 44100 Hz and there are 2 input channels (Stereo). My laptop does indeed have two built-in microphones. ‘s16le‘ means ‘signed 16-bit little-endian’.

4. Find out the audio card’s maximum sample rate (Hz):

$ arecord -f dat -r 60000 -D hw:0,0 -d 5 test.wav
Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 60000 Hz, Stereo
Warning: rate is not accurate (requested = 60000Hz, got = 48000Hz)
please, try the plug plugin
$

5. As the console output shows that my audio card supports a sample rate of 48000 Hz, edit /etc/pulse/daemon.conf and change the sample rate accordingly:

$ su
Password:
# grep sample-rate /etc/pulse/daemon.conf
; default-sample-rate = 44100
; alternate-sample-rate = 48000
# nano /etc/pulse/daemon.conf
# grep sample-rate /etc/pulse/daemon.conf
default-sample-rate = 48000
; alternate-sample-rate = 48000
# exit
exit
$

6. Find the buffer size and fragment size for each sound card:

$ echo autospawn = no >> ~/.config/pulse/client.conf
$ pulseaudio --kill
$ LANG=C timeout --foreground -k 10 -s kill 10 pulseaudio -vvvv 2>&1 | grep device.buffering -B 10
I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] sink.c: device.bus = "pci"
I: [pulseaudio] sink.c: device.vendor.id = "8086"
I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] sink.c: device.product.id = "3b56"
I: [pulseaudio] sink.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] sink.c: device.form_factor = "internal"
I: [pulseaudio] sink.c: device.string = "front:0"
I: [pulseaudio] sink.c: device.buffering.buffer_size = "19200"
I: [pulseaudio] sink.c: device.buffering.fragment_size = "4800"
--
I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] source.c: device.bus = "pci"
I: [pulseaudio] source.c: device.vendor.id = "8086"
I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] source.c: device.product.id = "3b56"
I: [pulseaudio] source.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] source.c: device.form_factor = "internal"
I: [pulseaudio] source.c: device.string = "front:0"
I: [pulseaudio] source.c: device.buffering.buffer_size = "19200"
I: [pulseaudio] source.c: device.buffering.fragment_size = "4800"
$ sed -i '$d' ~/.config/pulse/client.conf # You can just delete the file instead if it didn't exist in the first place.
$

N.B. Depending on your distribution, the PulseAudio file client.conf (if it exists) may be in a different sub-directory of the user’s home directory.

The console output shows that the buffer size was 19200 bits and the fragment size was 4800 bits.

7. Calculate the number of fragments and the fragment size (msec):

default-fragments = 19200 / 4800 = 4

default-fragments-size-msec

= device.buffering.fragment_size [bits] / (sample rate [Hz] x sample width [bits] x number of channels)

= 4800 / ( 48000 x 16 x 2 )

= 0.003125 seconds = 3.125 msec = 3 msec to the nearest integer

8. Edit /etc/pulse/daemon.conf to set these two parameters to the above-mentioned values:

# grep default-fragment /etc/pulse/daemon.conf
; default-fragments = 4
; default-fragment-size-msec = 25
# nano /etc/pulse/daemon.conf
# grep default-fragment /etc/pulse/daemon.conf
default-fragments = 4
default-fragment-size-msec = 3
# exit
exit
$

9. Restart PulseAudio:

$ pulseaudio --kill
$ pulseaudio --start # Only needed if you have 'autospawn = no' in ~/.config/pulse/client.conf
$

10. Check the PulseAudio configuration:

$ pulseaudio --dump-conf
### Read from configuration file: /etc/pulse/daemon.conf ###
daemonize = no
fail = yes
high-priority = yes
nice-level = -11
realtime-scheduling = yes
realtime-priority = 5
allow-module-loading = yes
allow-exit = yes
use-pid-file = yes
system-instance = no
local-server-type = user
cpu-limit = no
enable-shm = yes
flat-volumes = no
lock-memory = no
exit-idle-time = 20
scache-idle-time = 20
dl-search-path = /usr/lib64/pulse-5.0/modules
default-script-file = /etc/pulse/default.pa
load-default-script-file = yes
log-target =
log-level = notice
resample-method = auto
enable-remixing = yes
enable-lfe-remixing = no
default-sample-format = s16le
default-sample-rate = 48000
alternate-sample-rate = 48000
default-sample-channels = 2
default-channel-map = front-left,front-right
default-fragments = 4
default-fragment-size-msec = 3
enable-deferred-volume = yes
deferred-volume-safety-margin-usec = 8000
deferred-volume-extra-delay-usec = 0
shm-size-bytes = 0
log-meta = no
log-time = no
log-backtrace = 0
rlimit-fsize = -1
rlimit-data = -1
rlimit-stack = -1
rlimit-core = -1
rlimit-rss = -1
rlimit-as = -1
rlimit-nproc = -1
rlimit-nofile = 256
rlimit-memlock = -1
rlimit-locks = -1
rlimit-sigpending = -1
rlimit-msgqueue = -1
rlimit-nice = 31
rlimit-rtprio = 9
rlimit-rttime = 1000000
$

Notice that PulseAudio is now configured to use new values for default-sample-rate, default-fragments and default-fragment-size-msec.

$ echo autospawn = no >> ~/.config/pulse/client.conf
$ pulseaudio --kill
$ LANG=C timeout --foreground -k 10 -s kill 10 pulseaudio -vvvv 2>&1 | grep device.buffering -B 10
I: [pulseaudio] sink.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] sink.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] sink.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] sink.c: device.bus = "pci"
I: [pulseaudio] sink.c: device.vendor.id = "8086"
I: [pulseaudio] sink.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] sink.c: device.product.id = "3b56"
I: [pulseaudio] sink.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] sink.c: device.form_factor = "internal"
I: [pulseaudio] sink.c: device.string = "front:0"
I: [pulseaudio] sink.c: device.buffering.buffer_size = "2304"
I: [pulseaudio] sink.c: device.buffering.fragment_size = "576"
--
I: [pulseaudio] source.c: alsa.driver_name = "snd_hda_intel"
I: [pulseaudio] source.c: device.bus_path = "pci-0000:00:1b.0"
I: [pulseaudio] source.c: sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
I: [pulseaudio] source.c: device.bus = "pci"
I: [pulseaudio] source.c: device.vendor.id = "8086"
I: [pulseaudio] source.c: device.vendor.name = "Intel Corporation"
I: [pulseaudio] source.c: device.product.id = "3b56"
I: [pulseaudio] source.c: device.product.name = "5 Series/3400 Series Chipset High Definition Audio"
I: [pulseaudio] source.c: device.form_factor = "internal"
I: [pulseaudio] source.c: device.string = "front:0"
I: [pulseaudio] source.c: device.buffering.buffer_size = "2304"
I: [pulseaudio] source.c: device.buffering.fragment_size = "576"
$ rm ~/.config/pulse/client.conf # I didn't have a client.conf to begin with, so I just deleted it.
$ pulseaudio --start
$

Notice that PulseAudio is now using new values for device.buffering.buffer_size and device.buffering.fragment_size.

11. Check PulseAudio’s output sample rate:

$ pacmd list-sinks | grep sample
sample spec: s16le 2ch 48000Hz
$

Notice that PulseAudio is now using a new output sample rate.

12. Check PulseAudio’s input sample rate:

$ pacmd list-sources | grep sample
sample spec: s16le 2ch 48000Hz
sample spec: s16le 2ch 48000Hz
$

Notice that PulseAudio is now using a new input sample rate.

Audio in Linux becomes annoying again

At the moment I seem to be having more audio problems than usual. Last month I blogged about having to fix the ALSA Speaker volume level resetting to zero at boot, and recently two other audio problems have cropped up.

Thunderbird

I have been having trouble with Thunderbird’s ‘system sound’ that announces the arrival of a new e-mail. Lately, Thunderbird has started playing too loud and with significant distortion the audio clip it had been playing perfectly for the last four years. This is especially strange because I created that audio clip with Audacity from another audio clip that sounded too loud when Thunderbird played it. Ironically, the work-around for this latest problem was to switch to the original, much louder sound clip alert.wav instead of the quieter alert_quiet.wav. Not only does Thunderbird now play alert.wav at a lower volume than alert_quiet.wav, but the sound of alert.wav is not distorted when Thunderbird plays it. Yet if I play alert.wav and alert_quiet.wav using SMPlayer, the former is much louder than the latter and neither is distorted. Figure that one out.

The event notification sound that Thunderbird uses to remind me about an impending meeting scheduled in the calendar has now started sounding very distorted too. I still have not found a work-around for that. Event sounds played by the desktop environment I use (KDE) are not distorted, so what is Thunderbird doing? Perhaps the problem is not Thunderbird itself but the audio library it uses, so I need to investigate further.

Skype

Yet another audio problem cropped up this morning when I connected my laptop to an external monitor and keyboard (and thus I left the laptop’s lid almost closed) in an open-plan office and booted the laptop. I entered my username and password on the KDM log-in screen, and the KDE splash screen appeared as usual. After a few seconds the laptop’s speakers suddenly emitted a piercing, continuous howl; the well-known sound of audio feedback from speakers to microphone. It was LOUD. The volume control buttons on the keyboard made no difference, and the sound was so loud that everyone in the office noticed and several people came over to tell me to reset the BIOS (apparently that had fixed the problem for their laptops running Windows).

I kept my finger on the laptop’s power switch and, after several seconds, the laptop powered off. My laptop dual boots Windows 7 and Gentoo Linux, and the audio feedback did not occur when I booted Windows 7. After booting Linux again a couple of times and annoying everyone in the office even more, I discovered I could open the laptop’s lid far enough back to reduce the feedback to a low whine, so I could let KDE finish launching and display the Desktop. I then launched ALSAMixer and discovered that ‘Internal Mic Boost’ was set to 100%. So I immediately lowered it to zero. Then the penny dropped: I had used Skype the previous night without bothering to connect my headphones and external microphone, and Skype had automatically raised ‘Internal Mic Boost’ all the way up to 100%. So I immediately launched Skype, selected ‘Options’ > ‘Sound Devices’ and unticked ‘Allow Skype to automatically adjust my mixer levels’. The next thing I did was add the following lines to the script /etc/local.d/20set_alsa_volume.start mentioned in my previous blog post Fix for ALSA Speaker volume level resetting to zero at boot:

su -c "amixer -c 0 -- sset 'Internal Mic Boost' 0%" -s /bin/sh fitzcarraldo
su -c "amixer -c 0 -- sset 'Internal Mic' 0%" -s /bin/sh fitzcarraldo
su -c "amixer -c 0 -- sset 'Mic Boost' 0%" -s /bin/sh fitzcarraldo
su -c "amixer -c 0 -- sset 'Mic' 0%" -s /bin/sh fitzcarraldo

From now on, only I am allowed to adjust microphone settings! To avoid any possibility of feedback loops in future, the above-mentioned script sets all the microphone channels to zero and I will have to adjust them myself before use. I already have ALSAMixerGUI in the KDE Launcher menu, so it won’t be a big deal to do that.

This fiasco with Skype got me thinking: if Skype is set to automatically adjust mixer levels when you are in a conversation, when you exit Skype why doesn’t it automatically set mixer levels back to the way they were when Skype was launched? It could be done easily and would be more user-friendly than the current way Skype works.

Interrelationship between PulseAudio and ALSA

The final thing I did (yet again) was to adjust all the various ALSA channels and PulseAudio channels to try and get the resulting audio input and output sounding reasonable. This is easier said than done. I often have to mess around with ALSAMixer and PulseAudio Volume Control in order to get audio input and output working satisfactorily in all applications that use audio. It is actually more difficult than it sounds (ouch!) to get ALSA and PulseAudio ‘balanced’ (for want of a better word). In the days before PulseAudio existed, one only had to adjust ALSA. Now, with two agents controlling audio, the task turns out to be surprisingly awkward sometimes.

To sum up, boo to Thunderbird (or whatever it uses to play sounds), boo to Skype and boo to PulseAudio. I’m fed up with audio issues in Linux at the moment.😡

Update (January 19, 2015): It turns out that the problem in Thunderbird was due to PulseAudio. See my next post for details of how I fixed it.

Fix for ALSA Speaker volume level resetting to zero at boot

Up until 2011 the problem of the volume level in ALSA resetting to zero at boot was a common occurrence in my Linux installations. Actually it was a common occurrence in Linux, full stop; search the Web using keywords such as “alsa reset volume” and you’ll find umpteen links. In 2012 the situation improved and I thought the problem had become a thing of the past, but it resurfaced in 2013 on my main laptop and has plagued me through every update since (well, apart from in one release of KDE). Every time I reboot, the ALSA Speaker channel’s volume is zero when I log-in to KDE. And, as KMix is a PulseAudio mixer by default these days, I have to launch ALSAMixer and raise the volume of the Speaker channel manually.

I don’t know if the source of the problem lies in KDE, PulseAudio or ALSA itself. It did disappear after one upgrade to KDE earlier this year (I don’t recall which release of KDE) but returned in the next KDE upgrade, so I suspect it is a KDE issue. In earlier days I could resolve the problem using the commands alsactrl store and alsactl restore and similar approaches. However, this time none of those fixes work for me. The problem never bothered me much, as I always connect external speakers to my laptop’s headphone socket when I’m at home, and I don’t want my laptop emitting sounds at the office. Nevertheless, the fact the problem exists niggled me, so I decided to try and fix it. Rather than expending any more effort trying to get the usual approaches to work (the Web is littered with suggested fixes over the years), I decided to reset the Speaker volume to the same level at each boot by using an automatically-launched shell script. The method I use is given below, and should work with either OpenRC or systemd in Gentoo Linux.

1. Create the file /etc/local.d/20set_alsa_volume.start containing the following:

#!/bin/bash
# Reset ALSA Speaker channel on the first sound card to 90% after booting.
su -c "amixer -c 0 -- sset Speaker playback 90%" -s /bin/sh fitzcarraldo

(Replace “fitzcarraldo” with your own user name, of course.)

2. Make the script executable:

# chmod +x 20set_alsa_volume.start

That’s all there is to it. The volume of the ALSA Speaker channel is always set to 90% after I reboot and login to the desktop environment. KMix remembers the PulseAudio volume setting from the previous session, so I can still avoid blasting the laptop’s speakers.

By the way, the manual pages for the amixer and alsamixer commands make useful reading:

$ man amixer
$ man alsamixer

For example, the first audio card (Card 0) in my main laptop has the following controls:

$ amixer -c 0 scontrols
Simple mixer control ‘Master’,0
Simple mixer control ‘Headphone’,0
Simple mixer control ‘Speaker’,0
Simple mixer control ‘PCM’,0
Simple mixer control ‘Mic’,0
Simple mixer control ‘Mic Boost’,0
Simple mixer control ‘Beep’,0
Simple mixer control ‘Capture’,0
Simple mixer control ‘Auto-Mute Mode’,0
Simple mixer control ‘Digital’,0
Simple mixer control ‘Internal Mic’,0
Simple mixer control ‘Internal Mic Boost’,0

Update (January 29, 2015): I found the cause of the problem: PulseAudio. I edited the file /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf as per user agmg‘s January 2013 post Re: [SOLVED] Problems with PulseAudio 3.0 in the PCLinuxOS Forums:

Again, I had to edit the file: /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf

and change this:

[Element Speaker]
switch = mute
volume = off

to this:

[Element Speaker]
switch = mute
volume = merge

I realized that editing the [Element Desktop Speaker] section of the above file, has no effect on the issue. Only the edit to [Element Speaker] is needed in my case.

In my case this change forces the ALSA Speaker channel volume to 100% after rebooting (irrespective of the volume levels I set via ALSAMixer and KMix before shutdown).

The contents of the file /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf also include the following comment:

; See analog-output.conf.common for an explanation on the directives

The contents of the file analog-output.conf.common include the following comment:

; volume = ignore | merge | off | zero | <volume step> # What to do with this volume: ignore it, merge it into the device
;                                                      # volume slider, always set it to the lowest value possible, or always
;                                                      # set it to 0 dB (for whatever that means), or always set it to
;                                                      # <volume step> (this only makes sense in path configurations where
;                                                      # the exact hardware and driver are known beforehand).

So I could have tried volume = <volume step> instead of volume = merge, although I have no idea what value <volume step> would need to be. Anyway, the Bash script /etc/local.d/20set_alsa_volume.start that I created does the job in a different way without me having to tinker with the troublesome PulseAudio, so I reverted to volume = off in the file analog-output.conf and reverted to using /etc/local.d/20set_alsa_volume.start as explained earlier.

WebRTC – A viable alternative to Skype

webrtc_logoSkype for Linux 4.3 and upwards requires the use of PulseAudio, which has caused discontent amongst those Linux users who do not use PulseAudio. Although I do use PulseAudio, I recently found out about WebRTC, an API (application programming interface) for browser-based communication offering most of the functions provided by Skype, namely: voice calling, video chat, text chat, file sharing and screen sharing. The official WebRTC site states:

WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple JavaScript APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high quality, RTC applications to be developed in the browser via simple JavaScript APIs and HTML5.

WebRTC was originally released by Google but is now a draft standard of the World Wide Web Consortium, and is supported by Chrome, Firefox and Opera browsers. Several commercial Web sites offer WebRTC-based communications to fee-paying customers, but I thought I would try WebRTC by using one of the so-called ‘demo’ WebRTC pages. AppRTC is a WebRTC demo page which can be reached from a link on the official WebRTC site, but I prefer Multi-Party WebRTC Demo by TokBox which offers a more polished experience with better features. Both are free to use and viable substitutes to Skype for video chatting (one-to-one or conference).

So, how do you actually use WebRTC-based sites? Below is a quick guide to get you going.

Text and video chatting

Open the following URL in Chrome or Firefox:

https://opentokrtc.com/

Enter a Room Name that is likely to be unique. I used ‘fitzchat’ (without the quotes), but you can use any name you want.

The other party or parties can do the same thing, i.e. they enter the same Room Name as you, and you will all become connected.

Alternatively, to send an e-mail invitation to someone, click on the URL at the top of the pane on the right-hand side (which is Invite: https://opentokrtc.com/fitzchat in this example, as I chose to name the Room ‘fitzchat’). The partially visible pane at the right-hand side of the browser window will slide into full view when you click on it.

That’s all there is to it. You should see a video window showing each party, and they should see the same. Each party should also be able to hear the other parties. In the top right-hand corner of each video window is an icon (microphone for you; speaker for each of the other parties) which you can click on to mute/un-mute that party.

Click on the partially visible pane at the right-hand side of the browser window. Notice the ‘chat bar’ at the bottom where you enter commands and chat text. Read the grey instructions listed near the top of the pane:

Welcome to OpenTokRTC by TokBox
Type /nick your_name to change your name
Type /list to see list of users in the room
Type /help to see a list of commands
Type /hide to hide chat bar
Type /focus to lead the group
Type /unfocus to put everybody on equal standing

For example, to give myself a meaningful name instead of the default username Guest-0120e48c which was given to me automatically, I entered the following:

           /nick Fitz

Screen sharing

I found that screen sharing already works well in Chrome 36.0.1985.125 but is not yet supported in Firefox 31.0. It will be supported in Firefox 32 or 33, apparently, or you can already use Firefox Nightly providing you add the appropriate preferences via about:config.

To be able to share screens in Chrome, I had to perform two steps: enable a Chrome flag and install a Chrome extension. The two steps, which do not need to be repeated, are given below (see Ref. 1).

To enable screen sharing in Chrome, do the following:

  1. Open a new tab or window in Chrome.
  2. Copy the following link: chrome://flags/#enable-usermedia-screen-capture and paste it in the location bar.
  3. Click on the ‘Enable’ link below ‘Enable screen capture support in getUserMedia().’ at the very top of the screen.
  4. Click on the ‘Relaunch Now’ button at the bottom of the page to restart Chrome.

To install the screen sharing extension in Chrome, do the following:

  1. Launch Chrome and click on the Menu icon.
  2. Click on ‘Settings’.
  3. Click on ‘Extensions’.
  4. Click on ‘Get more extensions’ and search for ‘webrtc’.
  5. Download ‘WebRTC Desktop Sharing’.
  6. This places an icon to the right of the URL bar in Chrome.

To share your screen or just a window, do the following in Chrome:

  1. Click on the ‘Share Desktop’ icon to the right of the URL bar and select either ‘Screen’ or the window you wish to share.
  2. Click ‘Share’.
  3. When sharing has started in a new Chrome window, select the URL of the relevant tab in that window and send it to the other parties via the chat pane on the right-hand side of the first browser window.

To stop sharing, click on ‘Stop sharing’ and click on the ‘Share Desktop’ icon to the right of the URL bar to get it to return to displaying the ‘Share Desktop’ icon instead of the || (Pause) icon.

File sharing

I did not bother to try file sharing using WebRTC, but there are various Web sites you can use to do that. One such is ShareDrop, and googling will find others.

Caveats

Chrome 36.0.1985.125 and Firefox 31.0 were used in this trial (I did not try Opera). I found that video chat worked faultlessly when both parties were using Chrome, and when both parties were using Firefox. However, when one of the parties was using Firefox and the other was using Chrome, I could not see myself in one of the video boxes in the browser window (although I could see the other party in the other video box in the browser window). Furthermore, there was a grey bar across the middle of the video images in the AppRTC demo, whereas the Multi-Party WebRTC Demo video images were normal. Other than those two issues, the experience was smooth and straightforward. My recommendation would therefore be to use Multi-Party WebRTC Demo and for all the parties to use the same browser, be it Chrome or Firefox. If you want to share your screen or a window, the logical choice at the moment would be Chrome.

References

1 LiveMinutes Blog – Beta Testers: How To Activate Screen Sharing!

UPDATE (January 2, 2015): Mozilla has added a button to Firefox 34 to provide account-free video chat using WebRTC. Mozilla calls this feature ‘Firefox Hello’.

https://support.mozilla.org/en-US/kb/where-firefox-hello-button

I have it in Firefox 34.0.5 (I had to drag the ‘Hello’ button from ‘Customise’ | ‘Additional Tools and Features’). It works quite well. I didn’t bother creating an account; I just clicked on the ‘Email’ button to e-mail the automatically-generated URL to someone, and he clicked on the URL in the e-mail he received, which launched Firefox on his laptop and rang Firefox on my laptop. We tried both video and audio-only conversations, and both worked well. Firefox Hello is not as polished as Skype but, if Mozilla keeps working on it, they could end up with a good product.

Converting ape music files to mp3 in Linux

I had a file in the lossless ape (Monkey’s Audio) file format, and wanted to convert it to a .mp3 file so that I could play it on my portable mp3 player. As is usual in Linux, several alternative solutions exist, and I thought I’d try three of them for fun: shntool, ffmpeg and KDE’s Konvertible (Konvertible is a GUI for ffmpeg).

I already had ffmpeg and Konvertible installed, but not shntool. So first I installed shntool and the Monkey’s Audio codecs it uses:

# emerge media-sound/mac
# emerge media-sound/shntool

Here are the details of these two installed packages:

# eix -I shntool
[I] media-sound/shntool
Available versions: 3.0.10-r1 {alac flac mac shorten sox wavpack}
Installed versions: 3.0.10-r1(08:11:30 19/12/12)(flac -alac -mac -shorten -sox -wavpack)
Homepage: http://www.etree.org/shnutils/shntool/
Description: A multi-purpose WAVE data processing and reporting utility

# eix -I media-sound/mac
[I] media-sound/mac
Available versions: 3.99.4.5.7-r1^m {mmx static-libs}
Installed versions: 3.99.4.5.7-r1^m(07:52:12 19/12/12)(mmx -static-libs)
Homepage: http://etree.org/shnutils/shntool/
Description: Monkey's Audio Codecs

Then I used the following command to convert the file My Band 1971 CoolSounds.ape to mp3:

$ shntool conv -i ape -o 'cust ext=mp3 lame - %f' My\ Band\ 1971\ CoolSounds.ape
Converting [My Band 1971 CoolSounds.ape] (59:15.39) --> [My Band 1971 CoolSounds.mp3] : 100% OK
$

The KDE utility Konvertible was also able to convert it. I double-clicked on the file My Band 1971 CoolSounds.ape in Dolphin to launch Konvertible, selected libmp3lame in the ‘Codec:’ drop-down picklist, 192.00 kbits/s in the ‘Bitrate:’ drop-down picklist, clicked on the folder icon and selected /home/fitzcarraldo as the destination directory, and finally clicked ‘Convert’.

The mp3 files created by shntool and Konvertible were of different sizes:

File created by Konvertible:

$ file My\ Band\ 1971\ CoolSounds.mp3
My Band 1971 CoolSounds.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, Stereo
$ ls -la My\ Band\ 1971\ CoolSounds.mp3
-rw-r--r-- 1 fitzcarraldo users 85334024 Dec 19 08:11 My Band 1971 CoolSounds.mp3
$

File created by shntool:

$ file My\ Band\ 1971\ CoolSounds.mp3
My Band 1971 CoolSounds.mp3: MPEG ADTS, layer III, v1, 128 kbps, 44.1 kHz, JntStereo
$ ls -la My\ Band\ 1971\ CoolSounds.mp3
-rw-r--r-- 1 fitzcarraldo users 56889259 Dec 19 08:29 My Band 1971 CoolSounds.mp3
$

So I added the bitrate to the shntool command:

$ shntool conv -i ape -o 'cust ext=mp3 lame -b 192 - %f' My\ Band\ 1971\ CoolSounds.ape
Converting [My Band 1971 CoolSounds.ape] (59:15.39) --> [My Band 1971 CoolSounds.mp3] : 100% OK
$

and this time the mp3 file created by shntool is comparable to the mp3 file created by Konvertible:

$ file My\ Band\ 1971\ CoolSounds.mp3
My Band 1971 CoolSounds.mp3: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, JntStereo
$ ls -la My\ Band\ 1971\ CoolSounds.mp3
-rw-r--r-- 1 fitzcarraldo users 85333889 Dec 19 08:56 My Band 1971 CoolSounds.mp3
$

The ffmpeg command to do the same thing is:

$ ffmpeg -i My\ Band\ 1971\ CoolSounds.ape -ar 44100 -ab 192000 out.mp3
ffmpeg version 0.10.6 Copyright (c) 2000-2012 the FFmpeg developers
built on Nov 26 2012 07:06:40 with gcc 4.6.3
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --cxx=x86_64-pc-linux-gnu-g++ --ar=x86_64-pc-linux-gnu-ar --optflags='-O2 -march=native -pipe' --extra-cflags='-O2 -march=native -pipe' --extra-cxxflags='-O2 -march=native -pipe' --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --disable-vaapi --disable-vdpau --enable-runtime-cpudetect --enable-gnutls --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libdc1394 --enable-openal --disable-indev=v4l --disable-indev=oss --enable-x11grab --enable-libpulse --disable-outdev=oss --enable-libfreetype --enable-pthreads --enable-libgsm --enable-libspeex --disable-amd3dnow --disable-amd3dnowext --disable-altivec --disable-avx --disable-mmx2 --disable-ssse3 --disable-vis --disable-neon --cpu=ho
libavutil 51. 35.100 / 51. 35.100
libavcodec 53. 61.100 / 53. 61.100
libavformat 53. 32.100 / 53. 32.100
libavdevice 53. 4.100 / 53. 4.100
libavfilter 2. 61.100 / 2. 61.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 6.100 / 0. 6.100
libpostproc 52. 0.100 / 52. 0.100
Input #0, ape, from 'My Band 1971 CoolSounds.ape':
Metadata:
Album : CoolSounds
Title : C:\1\My Band 1971 CoolSounds
Comment : Exact Audio Copy
Duration: 00:59:15.47, start: 0.000000, bitrate: 829 kb/s
Stream #0:0: Audio: ape (APE / 0x20455041), 44100 Hz, stereo, s16
Output #0, mp3, to 'out.mp3':
Metadata:
TALB : CoolSounds
TIT2 : C:\1\My Band 1971 CoolSounds
Comment : Exact Audio Copy
TSSE : Lavf53.32.100
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 192 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (ape -> libmp3lame)
Press [q] to stop, [?] for help
size= 83334kB time=00:59:15.55 bitrate= 192.0kbits/s
video:0kB audio:83333kB global headers:0kB muxing overhead 0.000892%
$

and, as you can see below, the resulting mp3 file is the same size as the mp3 file created using Konvertible (not surprising, since Konvertible is a GUI front-end for ffmpeg) and virtually the same as the mp3 file created by shntool.

$ file out.mp3
out.mp3: Audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 192 kbps, 44.1 kHz, Stereo
$ ls -la out.mp3
-rw-r--r-- 1 fitzcarraldo users 85334024 Dec 20 18:14 out.mp3
$

So, there you have it: GUI or command line; take your pick!