If audio data is captured on another device (ie we are NOT using
loopback mode), then allow alsabat to analyze that data, by passing
a filename reference on the command line.
Add the '--readcapture' option to the argument parser. When
this option is specified, avoid doing a local capture, and instead
just read the audio data from the indicated file, and analyze that.
Fixes: https://github.com/alsa-project/alsa-utils/pull/166
Signed-off-by: Tim Bird <tim.bird@sony.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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>
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>
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>
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>
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>
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>
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>
Use dynamic temp file instead of fixed temp file to store recorded
wav data, for better security.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add common definitions of macros and data structures; Add functions
that used by multiple components, such as wav file reading and writing.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Bernard Gautier <bernard.gautier@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
