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First, normalize the bin power by dividing it by the number of input samples.
We need to multiply by 2 since we get half as many bins as input samples.
Second, check that low frequencies are under a given threshold. If there is a
pop or some noise, the low frequencies will be affected.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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This adds a snprintf-like behaviour to audio_extract_channel_s32_le.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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This function converts normalized doubles into an ALSA PCM format.
Instead of having per-format audio_signal_fill_* functions, we can only have
audio_signal_fill that outputs normalized doubles. Then in ALSA's playback
callback, we can simply use the new audio_convert_to function to fill the
buffer.
This makes the test code simpler and prevents code duplication when another
ALSA playback callback is implemented.
This adds a dependency of igt_audio over ALSA for the PCM format enum, but I
don't think this is a concern, since I don't see the point of using igt_audio
without igt_alsa. If this is an issue, it would always be possible to replace
ALSA's enum with our own in the future.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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This is less error-prone.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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- Make sure an audio signal isn't created with zero channels
- Make sure each channel has at least one frequency when synthesizing
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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Seems like I'm not used to the C89 style yet, and ASan caught it but with a
completely unrelated stack trace.
Signed-off-by: Simon Ser <simon.ser@intel.com>
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We need to use signed integers (hence "S" in the format name).
The reason this has gone unnoticed is that we were generating signals with half
of the amplitude they were supposed to have. As a result, using UINT32_MAX
(which is almost INT32_MAX * 2) resulted in the correct amplitude. In other
words, these two bugs were cancelling each other's effects.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Fixes: 42576d710255 ("lib/igt_audio: add support for S24_LE and S32_LE signal generation")
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This adds two new helpers to generate S24_LE and S32_LE signals.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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This is a safety net to ensure we haven't completely messed up our signal
generation. It would be unfortunate to send e.g. a null signal after a
refactoring.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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For each channel of the audio signal, we choose a set of frequencies to
generate. However we previously divided each sample by the total number of
frequencies, instead of the number of frequencies assigned to the channel we're
currently processing. This resulted in low amplitudes (0.5 instead of 1.0 for
2 channels).
This patch fixes this issue and sets the generated signal amplitude to 0.9.
Indeed, we want to detect if the receiver grows the signal, we don't want it
to get capped at 1.0 immediately.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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In order to be able to generate S32_LE signals, we need a higher precision than
S16_LE. Change the signal generation code to use double instead of int16_t and
introduce a new helper to fill a buffer with S16_LE samples. We already use
arrays of doubles elsewhere (see audio_signal_detect). Similar format-specific
helpers will be added for S32_LE in the future.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Arkadiusz Hiler <arkadiusz.hiler@intel.com>
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audio_signal_detect uses gsl_fft_real_radix2_transform which mutates the data
array. This can be surprising when calling audio_signal_detect and then read
again the data (e.g. for another check).
Instead of mutating the array, make audio_signal_detect less error-prone by
taking a const parameter. Do an internal copy before calling the gsl function.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
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audio_extract_channel_s32_le converts S32 samples to double. Prior to this
patch, the output samples weren't normalized, resulting in values up to
INT32_MAX.
Future amplitude tests need to be format-agnostic (ie. shouldn't care whether
the format used is S16 or S32). Thus, it is preferable to always use normalized
double values.
The current tests don't check the amplitude (or check it relative to the max.
on the whole buffer), so they still pass without any modification.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
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open(3) takes va_args after the flags and O_CREAT will read the first one. If
we don't provide one, this is undefined behaviour.
(Someone reported it broke the build for them)
Signed-off-by: Simon Ser <simon.ser@intel.com>
Fixes: 311baff151f9 ("tests/kms_chamelium: add dp-audio test")
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Signed-off-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
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This commit updates the audio test to make sure we receive a signal from both
audio channels. However this commit doesn't check that left and right channels
are not swapped. Such a check requires some more work (because the Chamelium
device does swap left and right channels) and will be implemented in a future
commit.
This commit adds a new channel argument to audio_signal_add_frequency, to add
a frequency to a single channel only.
Some light refactoring has been performed: a proper audio_signal_fini
function has been introduced and size_t in now used when it should be.
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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- Print matched audio devices
- Print min/max values when alsa_test_configuration fails
- Print debug log line when skipping a frequency
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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This new test ensures DisplayPort audio works by using the Chamelium.
It enables the DisplayPort output and sends an audio signal containing a set of
frequencies we choose to all HDMI/DisplayPort audio devices. It starts
recording audio on the Chamelium device and uses the stream server to retrieve
captured audio pages. It then checks that the capture audio signal contains the
frequencies we sent, and only those, by computing a FFT.
A new library has been added to libigt to communicate with the stream server.
It implements a simple custom TCP protocol.
In case the test fails, a WAV file with the captured data is saved on disk.
Right now the test has a few limitations:
- Only the first channel is checked
- IGT only generates audio with a single sampling rate (48 KHz)
- Audio data is not captured in real-time
These limitations will be lifted in future patches.
PulseAudio must not run during the tests since ALSA is used directly. To ensure
this, edit /etc/pulse/client.conf and add `autospawn=no`. Then run
`pulseaudio --kill`.
This commit deletes the existing audio tests. They weren't run and required an
exotic configuration (HDMI audio splitter, dummy HDMI sink and a line-in port
on the DUT).
This patch also changes lib/igt_audio to use uint16_t instead of short. The
rationale is:
- The standard says a short is at least 16 bit wide, but a short can be
larger (in practice it won't happen, but better use types correctly)
- It makes it clearer that the audio format is S16_LE, since "16" is
in the type name.
This patch depends on the following Chameleon bugs:
- https://crbug.com/948060
- https://crbug.com/950857
Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>
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It's not clear to the static analyzer that freq is assigned when
handling the previous frame and then used in the next. In fact, it
wasn't clear to me either before staring at the code for some
minutes. Initializing it to something makes both of us happier.
Signed-off-by: Petri Latvala <petri.latvala@intel.com>
Cc: Paul Kocialkowski <paul.kocialkowski@bootlin.com>
Reviewed-by: Paul Kocialkowski <paul.kocialkowski@bootlin.com>
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Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk>
Reviewed-by: Tvrtko Ursulin <tvrtko.ursulin@intel.com>
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Just a bit of OCD, I like it when connections within library modules
are a bit more obvious. igt.h is ok for tests, but let's use individual
include lines for libraries consistently.
Also order standard includes before igt ones.
v2: Rebase.
Acked-by: Jani Nikula <jani.nikula@intel.com>
Acked-by: Arkadiusz Hiler <arkadiusz.hiler@intel.com>
Acked-by: Petri Latvala <petri.latvala@intel.com>
Acked-by: Daniel Stone <daniels@collabora.com>
Acked-by: Radoslaw Szwichtenberg <radoslaw.szwichtenberg@intel.com>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
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This introduces an audio library, with dedicated helpers for both
generating signals and detecting peak frequencies in a signal.
This library paves the way for testing audio going through display
connectors, such as HDMI.
Signed-off-by: Paul Kocialkowski <paul.kocialkowski@linux.intel.com>
Reviewed-by: Lyude Paul <lyude@redhat.com>
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