[igt-dev] [PATCH i-g-t v3 09/10] tests/kms_chamelium: add a flatline audio test

[Ser, Simon]

On Tue, 2019-06-04 at 09:38 +0100, Tvrtko Ursulin wrote:
> On 27/05/2019 15:34, Simon Ser wrote:
> > This commit adds a flatline test alongside the existing frequencies test.
> > 
> > The test sends a constant value and checks that the amplitude is correct. A
> > window is used to check that each sample is within acceptable bounds. The test
> > is stopped as soon as 3 audio pages pass the test.
> > 
> > Signed-off-by: Simon Ser <simon.ser at intel.com>
> > Reviewed-by: Martin Peres <martin.peres at linux.intel.com>
> > ---
> >   tests/kms_chamelium.c | 101 ++++++++++++++++++++++++++++++++++++++++++
> >   1 file changed, 101 insertions(+)
> > 
> > diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
> > index 40ca93687c20..451a616f1a2e 100644
> > --- a/tests/kms_chamelium.c
> > +++ b/tests/kms_chamelium.c
> > @@ -772,6 +772,9 @@ test_display_frame_dump(data_t *data, struct chamelium_port *port)
> >   /* A streak of 3 gives confidence that the signal is good. */
> >   #define MIN_STREAK 3
> >   
> > +#define FLATLINE_AMPLITUDE 0.9 /* normalized, ie. in [0, 1] */
> 
> I assume the test is making triple sure it only ever outputs this signal 
> to connectors connected to Chamelium, in all possible scenarios? (I am 
> thinking it could be dangerous to some amps/speakers if by some kind of 
> accident.)

Not at all. The signal is sent to all HDMI/DP ports.

I have to check whether it's easy to match ALSA outputs to monitor
names.

Martin, is this a concern?

> Regards,
> 
> Tvrtko
> 
> > +#define FLATLINE_ACCURACY 0.001 /* ± 0.1% of the full amplitude */
> > +
> >   /* TODO: enable >48KHz rates, these are not reliable */
> >   static int test_sampling_rates[] = {
> >   	32000,
> > @@ -1138,6 +1141,103 @@ static bool test_audio_frequencies(struct audio_state *state)
> >   	return success;
> >   }
> >   
> > +static int audio_output_flatline_callback(void *data, void *buffer,
> > +					     int samples)
> > +{
> > +	struct audio_state *state = data;
> > +	double *tmp;
> > +	size_t len, i;
> > +
> > +	len = samples * state->playback.channels;
> > +	tmp = malloc(len * sizeof(double));
> > +	for (i = 0; i < len; i++)
> > +		tmp[i] = FLATLINE_AMPLITUDE;
> > +	audio_convert_to(buffer, tmp, len, state->playback.format);
> > +	free(tmp);
> > +
> > +	return state->run ? 0 : -1;
> > +}
> > +
> > +static bool detect_flatline_amplitude(double *buf, size_t buf_len)
> > +{
> > +	double min, max;
> > +	size_t i;
> > +	bool ok;
> > +
> > +	min = max = NAN;
> > +	for (i = 0; i < buf_len; i++) {
> > +		if (isnan(min) || buf[i] < min)
> > +			min = buf[i];
> > +		if (isnan(max) || buf[i] > max)
> > +			max = buf[i];
> > +	}
> > +
> > +	ok = (min >= FLATLINE_AMPLITUDE - FLATLINE_ACCURACY &&
> > +	      max <= FLATLINE_AMPLITUDE + FLATLINE_ACCURACY);
> > +	if (ok)
> > +		igt_debug("Flatline detected\n");
> > +	else
> > +		igt_debug("Flatline not detected (min=%f, max=%f)\n",
> > +			  min, max);
> > +	return ok;
> > +}
> > +
> > +static bool test_audio_flatline(struct audio_state *state)
> > +{
> > +	bool success;
> > +	int32_t *recv;
> > +	size_t recv_len, i, channel_len;
> > +	int streak, capture_chan;
> > +	double *channel;
> > +
> > +	alsa_register_output_callback(state->alsa,
> > +				      audio_output_flatline_callback, state,
> > +				      PLAYBACK_SAMPLES);
> > +
> > +	audio_state_start(state, "flatline");
> > +
> > +	recv = NULL;
> > +	recv_len = 0;
> > +	success = false;
> > +	while (!success && state->msec < AUDIO_TIMEOUT) {
> > +		audio_state_receive(state, &recv, &recv_len);
> > +
> > +		igt_debug("Detecting audio signal, t=%d msec\n", state->msec);
> > +
> > +		for (i = 0; i < state->playback.channels; i++) {
> > +			capture_chan = state->channel_mapping[i];
> > +			igt_assert(capture_chan >= 0);
> > +			igt_debug("Processing channel %zu (captured as "
> > +				  "channel %d)\n", i, capture_chan);
> > +
> > +			channel_len = audio_extract_channel_s32_le(NULL, 0,
> > +								   recv, recv_len,
> > +								   state->capture.channels,
> > +								   capture_chan);
> > +			channel = malloc(channel_len * sizeof(double));
> > +			audio_extract_channel_s32_le(channel, channel_len,
> > +						     recv, recv_len,
> > +						     state->capture.channels,
> > +						     capture_chan);
> > +
> > +			if (detect_flatline_amplitude(channel, channel_len))
> > +				streak++;
> > +			else
> > +				streak = 0;
> > +
> > +			free(channel);
> > +		}
> > +
> > +		success = streak == MIN_STREAK * state->playback.channels;
> > +	}
> > +
> > +	audio_state_stop(state, success);
> > +
> > +	free(recv);
> > +
> > +	return success;
> > +}
> > +
> >   static bool check_audio_configuration(struct alsa *alsa, snd_pcm_format_t format,
> >   				      int channels, int sampling_rate)
> >   {
> > @@ -1235,6 +1335,7 @@ test_display_audio(data_t *data, struct chamelium_port *port,
> >   			audio_state_init(&state, data, alsa, port,
> >   					 format, channels, sampling_rate);
> >   			success &= test_audio_frequencies(&state);
> > +			success &= test_audio_flatline(&state);
> >   			audio_state_fini(&state);
> >   
> >   			alsa_close_output(alsa);
> >