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

Martin Peres martin.peres at linux.intel.com
Mon May 27 10:20:28 UTC 2019


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 at 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 at 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);
>  		}
> 


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