[igt-dev] [PATCH i-g-t v2 8/9] tests/kms_chamelium: add pulse audio test

Simon Ser
2019-05-24T15:04:00Z

This commit adds a pulse test alongside the existing frequencies test.

The test sends an infinite pulse 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@intel.com>
---
tests/kms_chamelium.c | 100 ++++++++++++++++++++++++++++++++++++++++++
1 file changed, 100 insertions(+)

diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
index 14262831c3ff..073feff0d32d 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 PULSE_AMPLITUDE 0.9 /* normalized, ie. in [0, 1] */
+#define PULSE_ACCURACY 0.001 /* ± 0.1 % of the full amplitude */
+
/* TODO: enable >48KHz rates, these are not reliable */
static int test_sampling_rates[] = {
32000,
@@ -1136,6 +1139,102 @@ static bool test_audio_frequencies(struct audio_state *state)
return success;
}

+static int audio_output_pulse_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] = PULSE_AMPLITUDE;
+ audio_convert_to(buffer, tmp, len, state->playback.format);
+ free(tmp);
+
+ return state->run ? 0 : -1;
+}
+
+static bool detect_pulse_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 >= PULSE_AMPLITUDE - PULSE_ACCURACY &&
+ max <= PULSE_AMPLITUDE + PULSE_ACCURACY);
+ if (ok)
+ igt_debug("Pulse detected\n");
+ else
+ igt_debug("Pulse not detected (min=%f, max=%f)\n",
+ min, max);
+ return ok;
+}
+
+static bool test_audio_pulse(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_pulse_callback, state,
+ PLAYBACK_SAMPLES);
+
+ audio_state_start(state, "pulse");
+
+ 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_pulse_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)
{
@@ -1233,6 +1332,7 @@ 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_pulse(&state);
audio_state_fini(&state);

alsa_close_output(alsa);
--
2.21.0