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

Martin Peres
2019-05-27T12:46:06Z

The commit message is confusing because it might make people think this
has anything to do with pulseaudio.

Also, you are not really producing a pulse anyway. How about renaming it
square_wave, DC, or flatline?

On 24/05/2019 18:03, Simon Ser wrote:
> 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 */

No space before %.

Reviewed-by: Martin Peres <martin.peres@linux.intel.com>

> +
> /* 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);
>