Re: [igt-dev] [PATCH i-g-t v2 1/9] tests/kms_chamelium: refactor audio test

Martin Peres
2019-05-27T10:20:35Z

On 24/05/2019 18:03, Simon Ser wrote:
> Instead of shaving everything into do_test_display_audio, extract the logic to

shoving :)

> initialize, start, stop, finish an audio test in helper functions. The struct
> audio_state now carries all audio-related state.
>
> This will allow to easily add more audio tests that don't use sine waves (via
> audio_signal). This is necessary for future delay and amplitude tests.
>
> Signed-off-by: Simon Ser <simon.ser@intel.com>
> ---
> tests/kms_chamelium.c | 336 ++++++++++++++++++++++++------------------
> 1 file changed, 195 insertions(+), 141 deletions(-)
>
> diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
> index 8da6ec20759e..1a0a02ca2890 100644
> --- a/tests/kms_chamelium.c
> +++ b/tests/kms_chamelium.c
> @@ -812,17 +812,179 @@ static const snd_pcm_format_t test_formats[] = {
> static const size_t test_formats_count = sizeof(test_formats) / sizeof(test_formats[0]);
>
> struct audio_state {
> + struct alsa *alsa;
> + struct chamelium *chamelium;
> + struct chamelium_port *port;
> + struct chamelium_stream *stream;
> +
> + /* The capture format is only available after capture has started. */
> + struct {
> + snd_pcm_format_t format;
> + int channels;
> + int rate;
> + } playback, capture;
> +
> struct audio_signal *signal;
> - snd_pcm_format_t format;
> + int channel_mapping[8];
> +
> + int dump_fd;
> + char *dump_path;
> +
> + pthread_t thread;
> atomic_bool run;
> };
>
> +static void audio_state_init(struct audio_state *state, data_t *data,
> + struct alsa *alsa, struct chamelium_port *port,
> + snd_pcm_format_t format, int channels, int rate)
> +{
> + memset(state, 0, sizeof(*state));
> + state->dump_fd = -1;
> +
> + state->alsa = alsa;
> + state->chamelium = data->chamelium;
> + state->port = port;
> +
> + state->playback.format = format;
> + state->playback.channels = channels;
> + state->playback.rate = rate;
> +
> + alsa_configure_output(alsa, format, channels, rate);
> +
> + state->stream = chamelium_stream_init();
> + igt_assert(state->stream);
> +}
> +
> +static void audio_state_fini(struct audio_state *state)
> +{
> + chamelium_stream_deinit(state->stream);
> +}
> +
> +static void *run_audio_thread(void *data)
> +{
> + struct alsa *alsa = data;
> +
> + alsa_run(alsa, -1);
> + return NULL;
> +}
> +
> +static void audio_state_start(struct audio_state *state)
> +{
> + int ret;
> + bool ok;
> + size_t i, j;
> + enum chamelium_stream_realtime_mode stream_mode;
> + char dump_suffix[64];
> +
> + igt_debug("Starting test with playback format %s, sampling rate %d Hz "
> + "and %d channels\n",
> + snd_pcm_format_name(state->playback.format),
> + state->playback.rate, state->playback.channels);
> +
> + chamelium_start_capturing_audio(state->chamelium, state->port, false);
> +
> + stream_mode = CHAMELIUM_STREAM_REALTIME_STOP_WHEN_OVERFLOW;
> + ok = chamelium_stream_dump_realtime_audio(state->stream, stream_mode);
> + igt_assert(ok);
> +
> + /* Start playing audio */
> + state->run = true;
> + ret = pthread_create(&state->thread, NULL,
> + run_audio_thread, state->alsa);
> + igt_assert(ret == 0);
> +
> + /* The Chamelium device only supports this PCM format. */
> + state->capture.format = SND_PCM_FORMAT_S32_LE;
> +
> + /* Only after we've started playing audio, we can retrieve the capture
> + * format used by the Chamelium device. */
> + chamelium_get_audio_format(state->chamelium, state->port,
> + &state->capture.rate,
> + &state->capture.channels);

Is this a blocking call? If not, we have a race condition here...

> + if (state->capture.rate == 0) {
> + igt_debug("Audio receiver doesn't indicate the capture "
> + "sampling rate, assuming it's %d Hz\n",
> + state->playback.rate);
> + state->capture.rate = state->playback.rate;
> + }
> +
> + chamelium_get_audio_channel_mapping(state->chamelium, state->port,
> + state->channel_mapping);
> + /* Make sure we can capture all channels we send. */
> + for (i = 0; i < state->playback.channels; i++) {
> + ok = false;
> + for (j = 0; j < state->capture.channels; j++) {
> + if (state->channel_mapping[j] == i) {
> + ok = true;
> + break;
> + }
> + }
> + igt_assert(ok);
> + }

Nice check!

> +
> + if (igt_frame_dump_is_enabled()) {

Maybe you should rename this function, now that it is also used to dump
sound. How about igt_resource_dumping_is_enabled()? This does not have
to happen in this patch though.

This is a good patch:
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>

Martin

> + snprintf(dump_suffix, sizeof(dump_suffix),
> + "capture-%s-%dch-%dHz",
> + snd_pcm_format_name(state->playback.format),
> + state->playback.channels, state->playback.rate);
> +
> + state->dump_fd = audio_create_wav_file_s32_le(dump_suffix,
> + state->capture.rate,
> + state->capture.channels,
> + &state->dump_path);
> + igt_assert(state->dump_fd >= 0);
> + }
> +}
> +
> +static void audio_state_stop(struct audio_state *state, bool success)
> +{
> + bool ok;
> + int ret;
> + struct chamelium_audio_file *audio_file;
> +
> + igt_debug("Stopping audio playback\n");
> + state->run = false;
> + ret = pthread_join(state->thread, NULL);
> + igt_assert(ret == 0);
> +
> + ok = chamelium_stream_stop_realtime_audio(state->stream);
> + igt_assert(ok);
> +
> + audio_file = chamelium_stop_capturing_audio(state->chamelium,
> + state->port);
> + if (audio_file) {
> + igt_debug("Audio file saved on the Chamelium in %s\n",
> + audio_file->path);
> + chamelium_destroy_audio_file(audio_file);
> + }
> +
> + if (state->dump_fd >= 0) {
> + close(state->dump_fd);
> + state->dump_fd = -1;
> +
> + if (success) {
> + /* Test succeeded, no need to keep the captured data */
> + unlink(state->dump_path);
> + } else
> + igt_debug("Saved captured audio data to %s\n",
> + state->dump_path);
> + free(state->dump_path);
> + state->dump_path = NULL;
> + }
> +
> + igt_debug("Audio test result for format %s, sampling rate %d Hz and "
> + "%d channels: %s\n",
> + snd_pcm_format_name(state->playback.format),
> + state->playback.rate, state->playback.channels,
> + success ? "ALL GREEN" : "FAILED");
> +}
> +
> static int
> audio_output_callback(void *data, void *buffer, int samples)
> {
> struct audio_state *state = data;
>
> - switch (state->format) {
> + switch (state->playback.format) {
> case SND_PCM_FORMAT_S16_LE:
> audio_signal_fill_s16_le(state->signal, buffer, samples);
> break;
> @@ -839,55 +1001,19 @@ audio_output_callback(void *data, void *buffer, int samples)
> return state->run ? 0 : -1;
> }
>
> -static void *
> -run_audio_thread(void *data)
> +static bool do_test_display_audio(struct audio_state *state)
> {
> - struct alsa *alsa = data;
> -
> - alsa_run(alsa, -1);
> - return NULL;
> -}
> -
> -static bool
> -do_test_display_audio(data_t *data, struct chamelium_port *port,
> - struct alsa *alsa, snd_pcm_format_t playback_format,
> - int playback_channels, int playback_rate)
> -{
> - int ret, capture_rate, capture_channels, msec, freq, step;
> - struct chamelium_audio_file *audio_file;
> - struct chamelium_stream *stream;
> - enum chamelium_stream_realtime_mode stream_mode;
> - struct audio_signal *signal;
> + int msec, freq, step;
> int32_t *recv, *buf;
> double *channel;
> size_t i, j, streak, page_count;
> size_t recv_len, buf_len, buf_cap, buf_size, channel_len;
> bool ok, success;
> - char dump_suffix[64];
> - char *dump_path = NULL;
> - int dump_fd = -1;
> - pthread_t thread;
> - struct audio_state state = {};
> - int channel_mapping[8], capture_chan;
> -
> - igt_debug("Testing with playback format %s, sampling rate %d Hz and "
> - "%d channels\n",
> - snd_pcm_format_name(playback_format),
> - playback_rate, playback_channels);
> - alsa_configure_output(alsa, playback_format,
> - playback_channels, playback_rate);
> + int capture_chan;
>
> - chamelium_start_capturing_audio(data->chamelium, port, false);
> -
> - stream = chamelium_stream_init();
> - igt_assert(stream);
> -
> - stream_mode = CHAMELIUM_STREAM_REALTIME_STOP_WHEN_OVERFLOW;
> - ok = chamelium_stream_dump_realtime_audio(stream, stream_mode);
> - igt_assert(ok);
> -
> - signal = audio_signal_init(playback_channels, playback_rate);
> - igt_assert(signal);
> + state->signal = audio_signal_init(state->playback.channels,
> + state->playback.rate);
> + igt_assert(state->signal);
>
> /* We'll choose different frequencies per channel to make sure they are
> * independent from each other. To do so, we'll add a different offset
> @@ -900,62 +1026,21 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> * later on. We cannot retrieve the capture rate before starting
> * playing audio, so we don't really have the choice.
> */
> - step = 2 * playback_rate / CAPTURE_SAMPLES;
> + step = 2 * state->playback.rate / CAPTURE_SAMPLES;
> for (i = 0; i < test_frequencies_count; i++) {
> - for (j = 0; j < playback_channels; j++) {
> + for (j = 0; j < state->playback.channels; j++) {
> freq = test_frequencies[i] + j * step;
> - audio_signal_add_frequency(signal, freq, j);
> + audio_signal_add_frequency(state->signal, freq, j);
> }
> }
> - audio_signal_synthesize(signal);
> + audio_signal_synthesize(state->signal);
>
> - state.signal = signal;
> - state.format = playback_format;
> - state.run = true;
> - alsa_register_output_callback(alsa, audio_output_callback, &state,
> + alsa_register_output_callback(state->alsa, audio_output_callback, state,
> PLAYBACK_SAMPLES);
>
> - /* Start playing audio */
> - ret = pthread_create(&thread, NULL, run_audio_thread, alsa);
> - igt_assert(ret == 0);
> + audio_state_start(state);
>
> - /* Only after we've started playing audio, we can retrieve the capture
> - * format used by the Chamelium device. */
> - chamelium_get_audio_format(data->chamelium, port,
> - &capture_rate, &capture_channels);
> - if (capture_rate == 0) {
> - igt_debug("Audio receiver doesn't indicate the capture "
> - "sampling rate, assuming it's %d Hz\n", playback_rate);
> - capture_rate = playback_rate;
> - } else
> - igt_assert(capture_rate == playback_rate);
> -
> - chamelium_get_audio_channel_mapping(data->chamelium, port,
> - channel_mapping);
> - /* Make sure we can capture all channels we send. */
> - for (i = 0; i < playback_channels; i++) {
> - ok = false;
> - for (j = 0; j < capture_channels; j++) {
> - if (channel_mapping[j] == i) {
> - ok = true;
> - break;
> - }
> - }
> - igt_assert(ok);
> - }
> -
> - if (igt_frame_dump_is_enabled()) {
> - snprintf(dump_suffix, sizeof(dump_suffix),
> - "capture-%s-%dch-%dHz",
> - snd_pcm_format_name(playback_format),
> - playback_channels, playback_rate);
> -
> - dump_fd = audio_create_wav_file_s32_le(dump_suffix,
> - capture_rate,
> - capture_channels,
> - &dump_path);
> - igt_assert(dump_fd >= 0);
> - }
> + igt_assert(state->capture.rate == state->playback.rate);
>
> /* Needs to be a multiple of 128, because that's the number of samples
> * we get per channel each time we receive an audio page from the
> @@ -970,7 +1055,7 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> channel_len = CAPTURE_SAMPLES;
> channel = malloc(sizeof(double) * channel_len);
>
> - buf_cap = capture_channels * channel_len;
> + buf_cap = state->capture.channels * channel_len;
> buf = malloc(sizeof(int32_t) * buf_cap);
> buf_len = 0;
>
> @@ -982,7 +1067,7 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> msec = 0;
> i = 0;
> while (!success && msec < AUDIO_TIMEOUT) {
> - ok = chamelium_stream_receive_realtime_audio(stream,
> + ok = chamelium_stream_receive_realtime_audio(state->stream,
> &page_count,
> &recv, &recv_len);
> igt_assert(ok);
> @@ -994,26 +1079,27 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> continue;
> igt_assert(buf_len == buf_cap);
>
> - if (dump_fd >= 0) {
> + if (state->dump_fd >= 0) {
> buf_size = buf_len * sizeof(int32_t);
> - igt_assert(write(dump_fd, buf, buf_size) == buf_size);
> + igt_assert(write(state->dump_fd, buf, buf_size) == buf_size);
> }
>
> - msec = i * channel_len / (double) capture_rate * 1000;
> + msec = i * channel_len / (double) state->capture.rate * 1000;
> igt_debug("Detecting audio signal, t=%d msec\n", msec);
>
> - for (j = 0; j < playback_channels; j++) {
> - capture_chan = channel_mapping[j];
> + for (j = 0; j < state->playback.channels; j++) {
> + capture_chan = state->channel_mapping[j];
> igt_assert(capture_chan >= 0);
> igt_debug("Processing channel %zu (captured as "
> "channel %d)\n", j, capture_chan);
>
> audio_extract_channel_s32_le(channel, channel_len,
> buf, buf_len,
> - capture_channels,
> + state->capture.channels,
> capture_chan);
>
> - if (audio_signal_detect(signal, capture_rate, j,
> + if (audio_signal_detect(state->signal,
> + state->capture.rate, j,
> channel, channel_len))
> streak++;
> else
> @@ -1023,49 +1109,15 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
> buf_len = 0;
> i++;
>
> - success = streak == MIN_STREAK * playback_channels;
> + success = streak == MIN_STREAK * state->playback.channels;
> }
>
> - igt_debug("Stopping audio playback\n");
> - state.run = false;
> - ret = pthread_join(thread, NULL);
> - igt_assert(ret == 0);
> -
> - alsa_close_output(alsa);
> -
> - igt_debug("Audio test result for format %s, sampling rate %d Hz and "
> - "%d channels: %s\n",
> - snd_pcm_format_name(playback_format),
> - playback_rate, playback_channels,
> - success ? "ALL GREEN" : "FAILED");
> -
> - if (dump_fd >= 0) {
> - close(dump_fd);
> - if (success) {
> - /* Test succeeded, no need to keep the captured data */
> - unlink(dump_path);
> - } else
> - igt_debug("Saved captured audio data to %s\n", dump_path);
> - free(dump_path);
> - }
> + audio_state_stop(state, success);
>
> free(recv);
> free(buf);
> free(channel);
> -
> - ok = chamelium_stream_stop_realtime_audio(stream);
> - igt_assert(ok);
> -
> - audio_file = chamelium_stop_capturing_audio(data->chamelium,
> - port);
> - if (audio_file) {
> - igt_debug("Audio file saved on the Chamelium in %s\n",
> - audio_file->path);
> - chamelium_destroy_audio_file(audio_file);
> - }
> -
> - audio_signal_fini(signal);
> - chamelium_stream_deinit(stream);
> + audio_signal_fini(state->signal);
>
> return success;
> }
> @@ -1112,6 +1164,7 @@ test_display_audio(data_t *data, struct chamelium_port *port,
> int fb_id, i, j;
> int channels, sampling_rate;
> snd_pcm_format_t format;
> + struct audio_state state;
>
> igt_require(alsa_has_exclusive_access());
>
> @@ -1161,9 +1214,10 @@ test_display_audio(data_t *data, struct chamelium_port *port,
>
> run = true;
>
> - success &= do_test_display_audio(data, port, alsa,
> - format, channels,
> - sampling_rate);
> + audio_state_init(&state, data, alsa, port,
> + format, channels, sampling_rate);
> + success &= do_test_display_audio(&state);
> + audio_state_fini(&state);
>
> alsa_close_output(alsa);
> }
>