The code releasing the each card object may access to NULL when a
bogus count is given. Add a NULL check just to make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define a USE_RST2MAN conditional so that, when available, rst2man can be
used to generate man pages from reStructuredText source files.
The code follows what is done to check for xmlto.
On Debian system, the rst2man executable is provided by python-docutils
or python3-docutils.
Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The same $xmlto variable is used both in AC_ARG_ENABLE and
AC_CHECK_PROG, but the latter is not setting a value to it when the
program is not found.
These two facts result in the "yes" value from the AC_ARG_ENABLE macro
to be still kept in the variable when the program is not found by
AC_CHECK_PROG, causing USE_XMLTO to be always set, finally resulting in
a build failure in case the xmlto program is not actually in the PATH.
As possible fix could have been to set "no" as a value in AC_CHECK_PROG
when program is not found.
However using two separate variables is more explicit, so fix the issue
this way.
Signed-off-by: Antonio Ospite <ao2@ao2.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The channel selection currently does not work properly when a channel
map control is provided but no manual channel map was explicitly
requested with "-m".
For example, the CEA/HDMI 6ch (surround 5.1) map is:
FL, FR, LFE, FC, RL, RR.
Tested command: speaker-test -D hdmi -c 6 -t wav
Speaker-test tries to play channels in this following order:
0 - Front Left
3 - Front Center
1 - Front Right
5 - Rear Right
4 - Rear Left
2 - LFE
But wav file played on associated speakers are not aligned. Here are
the real files played:
0- /usr/share/sounds/alsa/Front_Left.wav => OK
3- /usr/share/sounds/alsa/Rear_Right.wav => OK
1- /usr/share/sounds/alsa/Front_Right.wav => OK
5- /usr/share/sounds/alsa/Rear_Center.wav => KO
4- found file /usr/share/sounds/alsa/Front_Center.wav => KO
2- /usr/share/sounds/alsa/Rear_Left.wav => KO
Issue is that associated wav files ordering is reworked only if
channel_map_set variable is set.
Fix consists in allowing wavs re-ordering if a channel mapping as been
get or set, i.e. channel_map is not null.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If PCM is paused and then we do system supend-resume, the stream throws
error(EBADF) when stream is paused released.
Check the pcm state before pause/release and if stream is suspended,
call snd_pcm_resume to resume the stream.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new option to amidi tool: sysex-interval.
It adds a delay (in milliseconds) in between each SysEx message - it searches
for a 0xF7 byte.
This is very useful when sending firmware updates to a remote device via SysEx
or any other use that requires this delay in between SysEx messages.
`amidi' manual was updated with an example usage as well.
Signed-off-by: Felipe F. Tonello <eu@felipetonello.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support audio pause/resume for playback and capture. The user can
pause alsabat playback/capture threads by sending a signal. The patch
provides a method for QA to quick test audio during system s3.
Signed-off-by: Keqiao, Zhang <keqiao.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alsabat reports -86 error when system suspend and resume. Check the
return value of read_to_pcm() and write_to_pcm(), when -x8 err is
detected, do resume and wait for read/write to pcm to complete.
Write PCM device error: Streams pipe error(-86)
Read PCM device error: Streams pipe error(-86)
*** Error in alsabat: double free or corruption (out): 0x00007fb438001810 ***
Signed-off-by: Keqiao, Zhang <keqiao.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch includes the reference asound.state config files
on Intel Skylake, Broadwell and Hsawell platforms
Signed-off-by: Focus Luo <focus.luo@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch includes automated test scripts for linux audio driver
based on alsa-lib interface by using alsabat as test tool.
It supports analog and display(HDMI/DP) audio test.
The package needs the alsa-utils, alsa-lib installed environment.
Signed-off-by: Focus Luo <focus.luo@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timeout is not supposed to expire when ignored messages are
received. This cannot be handled with the poll() timeout, so add
a separate timer.
Reported-by: Martin Tarenskeen <m.tarenskeen@gmail.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Active Sensing messages are sent by many devices in the background and
would only interfere with the actual messages that amidi is supposed to
capture. Therefore, amidi ignores them by default. However, there are
also devices that send Clock messages with the same problem, so it is
a better idea to filter them out, too.
Reported-by: Martin Tarenskeen <m.tarenskeen@gmail.com>
Reviewd-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The compilation fails due to multiple defination of snr_is_valid
common.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here
signal.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here
latencytest.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here
analyze.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here
alsa.o: In function `snr_is_valid':
bat/common.h:99: multiple definition of `snr_is_valid'
bat.o:bat/common.h:99: first defined here
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alsabat reports error when noise above threshold be detected.
Use either of the options below to designate the threshold. (e.g.
if the ratio of noise to signal is 5%, the snr is about 26dB.)
--snr-db <value in dB>
--snr-pc <value in %>
The noise detection is performed in time domain. On each period
of the sine wave being analyzed, alsabat substracts a clean sine
wave from the source, calculates the RMS value of the residual,
and compares the result with the threshold. At last, alsabat
returns the number of periods with noise above threshold. 0 is
returned when the source is clean.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add function generate_sine_wave_raw_mono(). It serves as a single
channel sine wave generator, to provide data for calculation (e.g.
for noise analysis).
The function is similar to generate_sine_wave(), but a lite revision.
It has no dependency on bat channels and target frequency, no malloc
inside, no data conversion from float to integer samples, and supports
one channel only.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function adjust_waveform() is a component of generate_sine_wave(),
and depended on bat->channels parameter. Add parameter "channels" to
remove the dependency, and then adjust_waveform() can be applied on
other use cases, e.g. a single channel sine wave generator.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Aligning the data type of fftw analyzer, sample converter and other
components on float, because:
1. avoid unnecessary data type conversion;
2. using float is more efficient than using double;
3. the extra double accuracy is not required.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio latency is the time delay as an audio signal passes through
a system. There are many kinds of audio latency metrics. One useful
metric is the round trip latency, which is the sum of output latency
and input latency.
The measurement step works like below:
1. Listen and measure the average loudness of the environment for
one second;
2. Create a threshold value 16 decibels higher than the average
loudness;
3. Begin playing a ~1000 Hz sine wave and start counting the samples
elapsed;
4. Stop counting and playing if the input's loudness is higher than
the threshold, as the output wave is probably coming back;
5. Calculate the audio latency value in milliseconds.
Signed-off-by: Zhang Vivian <vivian.zhang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a possible memory leak in generate_sine_wave(). Memory free was
ignored when the function return an error.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add buffer size and period size settings in alsabat.
With -E and -B options, alsabat performs the test with
specified buffer size and period size
Signed-off-by: Zhang Vivian <vivian.zhang@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In loopback test, alsabat use pthread_join(pthread_t thread, **retval)
to wait for the capture thread to terminate. If the capture thread was
canceled, PTHREAD_CANCELED is placed in *retval, and the access to the
**retval will fail. Add status check to prevent illegal access to the
**retval.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If -d was given to arecord while commit
8aa13eec80 (now reverted) was in effect,
the last read would be shorter than the chunk size, but pcm_read would
read and return the chunk size, the samples were discarded, and
capture() continued in a loop because count never reached 0. arecord
opens a new file each loop iteration, if arecord is dynamically naming
files, --use-strftime option or beyond the wave 2GB limit, this will
generate a series of header only wave files. If the file is unique
the originally recorded data is lost and it will continue overwriting
the same file with a header only wave file.
While the current pcm_read can't fail (it can exit), it is better to
just fix this lurking bug in case it is "fixed" again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use "configure --enable-alsabat-backend-tiny" for alsabat to use
tinyalsa as backend lib. On a system that has both ALSA and tinyalsa
installed, alsabat will use ALSA library by default.
The intention is for alsabat to run on tinyalsa platforms such as
Android or some Internet of Things(IoT) devices.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move all alsa callings to a single block (alsa.c), so other blocks
such as the main structure, the signal process and the data analysis
modules will be independent to alsa, and new modules such as a
tinyalsa interface can be easily embedded into alsabat.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use general data generator to replace local function, so other
modules can reuse the data generator rather than re-implement it.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unnecessary prints in playback and capture threads, and replace
the return value "0" with error code for convenience of maintaining.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In playback thread, use general function update_wav_header()
to replace a bunch of code, so the structure is cleaner and
no need to define variable "wav".
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactoring ALSA capture thread:
1. Move file open/seek operations to sub function, so all file
processes are now on a single function (read_from_pcm_loop()), so
the structure is more reasonable, the function API is simplified
and no need file cleanup in thread loop.
2. Replace the wav header processing lines with a general function
(update_wav_header()), which can be reused in other sections.
3. Add pthread_exit() for thread to exit safely in single line mode,
and correct comment.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use variable instead of 0/1 to indicate the return value of
playback and capture threads.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All files should be opened in either "rb" or "wb" in current
usage.
Remove incorrect and unneccesary prints.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Preserve errno value before use, since the value might be
changed by another library call.
Add "#include <errno.h>" and remove redundant include.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add bash script for alsabat feature test. It covers basic alsabat
features, including waveform and wav file generate, playback,
capture and analysis with configurable parameters under different
work modes. (loopback, single line, standalone and local mode)
Usage examples:
alsabat-test.sh
alsabat-test.sh plughw:1,0
alsabat-test.sh plughw:1,0 plughw:1,0
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for standalone mode where alsabat will run on a
different machine to the one being tested.
In standalone mode, the alsabat just generates, playback and
capture sound data like in normal mode, but does not analyze.
The alsabat being built without libfftw3 support is always work
in standalone mode.
The alsabat in normal mode can also bypass data analysis using
option "--standalone".
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add default name for the playback and capture devices, in case
they were not set by user through '-D', '-P' or '-C' options.
Previously, if no device be specified, the alsabat will start
a playback thread and a capture thread, and then exit the
threads with error log.
If only one of playback and capture is specified, the alsabat
will work on single line mode as before, where only one thread
(playback or capture) will be started.
The patch was tested on Ubuntu and Chrome OS.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
when a user enters a command, he expects his command to be executed
as specified or aborted if it is impossible to fulfill his request
right now a command like "arecord -d hw:1,0 --max-file-time 1h recording.wav"
will happily record something and exit without error status while the resulting
recording contains definitely not what the user requested
to fix this, the patch handles the number parsing function's error channel
and checks whether the parsed number has any trailing characters
Signed-off-by: erwin <nebelbank@posteo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
rawmidi devices expose the card number via IOCTLs, which allows to
find the corresponding device in sysfs.
The sequencer provides no identifing data. Chromium works around this
issue by scanning rawmidi as well as sequencer devices and matching
them by using assumtions, how the kernel register sequencer devices.
This patch adds support for displaying the sound card number/PID to
aconnect.
Signed-off-by: Martin Koegler <martin.koegler@chello.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Truncate the sample frames to powers of 2, since the FFTW algorithm
runs especially fast in this case, and other sizes may be computed
by means of a slow, general-purpose algorithm.
In my test environment applying the patch, a sound clip of 33072
frames is cut off to 32768 frames before analysis, and the time
cost is reduced from 6.128s to 0.224s.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
