tests/kms_chamelium: add pulse test

index cfdf7617a65a..75f03d8469aa 100644
@@ -1020,7 +1020,7 @@ bool chamelium_has_audio_support(struct chamelium *chamelium,
  */
 void chamelium_get_audio_channel_mapping(struct chamelium *chamelium,
 					 struct chamelium_port *port,
-					 int mapping[static 8])
+					 int mapping[static CHAMELIUM_MAX_AUDIO_CHANNELS])
 {
 	xmlrpc_value *res, *res_channel;
 	int res_len, i;

@@ -1028,7 +1028,7 @@ void chamelium_get_audio_channel_mapping(struct chamelium *chamelium,
 	res = chamelium_rpc(chamelium, port, "GetAudioChannelMapping", "(i)",
 			    port->id);
 	res_len = xmlrpc_array_size(&chamelium->env, res);
-	igt_assert(res_len == 8);
+	igt_assert(res_len == CHAMELIUM_MAX_AUDIO_CHANNELS);
 	for (i = 0; i < res_len; i++) {
 		xmlrpc_array_read_item(&chamelium->env, res, i, &res_channel);
 		xmlrpc_read_int(&chamelium->env, res_channel, &mapping[i]);

@@ -1058,8 +1058,10 @@ static void audio_format_from_xml(struct chamelium *chamelium,
 
 	if (rate)
 		xmlrpc_read_int(&chamelium->env, res_rate, rate);
-	if (channels)
+	if (channels) {
 		xmlrpc_read_int(&chamelium->env, res_channel, channels);
+		igt_assert(*channels <= CHAMELIUM_MAX_AUDIO_CHANNELS);
+	}
 
 	xmlrpc_DECREF(res_channel);
 	xmlrpc_DECREF(res_sample_format);

index 33362b266ce4..28f726c91e83 100644
@@ -65,6 +65,12 @@ struct chamelium_audio_file {
  */
 #define CHAMELIUM_DEFAULT_EDID 0
 
+/**
+ * CHAMELIUM_MAX_AUDIO_CHANNELS: the maximum number of audio capture channels
+ * supported by Chamelium.
+ */
+#define CHAMELIUM_MAX_AUDIO_CHANNELS 8
+
 struct chamelium *chamelium_init(int drm_fd);
 void chamelium_deinit(struct chamelium *chamelium);
 void chamelium_reset(struct chamelium *chamelium);

@@ -116,7 +122,7 @@ bool chamelium_has_audio_support(struct chamelium *chamelium,
 				 struct chamelium_port *port);
 void chamelium_get_audio_channel_mapping(struct chamelium *chamelium,
 					 struct chamelium_port *port,
-					 int mapping[static 8]);
+					 int mapping[static CHAMELIUM_MAX_AUDIO_CHANNELS]);
 void chamelium_get_audio_format(struct chamelium *chamelium,
 				struct chamelium_port *port,
 				int *rate, int *channels);

index 8da6ec20759e..27693681728e 100644
@@ -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[CHAMELIUM_MAX_AUDIO_CHANNELS];
+
+	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);
+	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);
+	}
+
+	if (igt_frame_dump_is_enabled()) {
+		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);
 		}

index 27693681728e..71489e5df9d7 100644
@@ -827,6 +827,9 @@ struct audio_state {
 	struct audio_signal *signal;
 	int channel_mapping[CHAMELIUM_MAX_AUDIO_CHANNELS];
 
+	size_t recv_pages;
+	int msec;
+
 	int dump_fd;
 	char *dump_path;
 

@@ -876,6 +879,9 @@ static void audio_state_start(struct audio_state *state)
 	enum chamelium_stream_realtime_mode stream_mode;
 	char dump_suffix[64];
 
+	state->recv_pages = 0;
+	state->msec = 0;
+
 	igt_debug("Starting test with playback format %s, sampling rate %d Hz "
 		  "and %d channels\n",
 		  snd_pcm_format_name(state->playback.format),

@@ -936,6 +942,29 @@ static void audio_state_start(struct audio_state *state)
 	}
 }
 
+static void audio_state_receive(struct audio_state *state,
+				int32_t **recv, size_t *recv_len)
+{
+	bool ok;
+	size_t page_count;
+	size_t recv_size;
+
+	ok = chamelium_stream_receive_realtime_audio(state->stream,
+						     &page_count,
+						     recv, recv_len);
+	igt_assert(ok);
+
+	state->msec = state->recv_pages * *recv_len
+		      / (double) state->capture.channels
+		      / (double) state->capture.rate * 1000;
+	state->recv_pages++;
+
+	if (state->dump_fd >= 0) {
+		recv_size = *recv_len * sizeof(int32_t);
+		igt_assert(write(state->dump_fd, *recv, recv_size) == recv_size);
+	}
+}
+
 static void audio_state_stop(struct audio_state *state, bool success)
 {
 	bool ok;

@@ -1003,12 +1032,12 @@ audio_output_callback(void *data, void *buffer, int samples)
 
 static bool do_test_display_audio(struct audio_state *state)
 {
-	int msec, freq, step;
+	int 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;
+	size_t i, j, streak;
+	size_t recv_len, buf_len, buf_cap, channel_len;
+	bool success;
 	int capture_chan;
 
 	state->signal = audio_signal_init(state->playback.channels,

@@ -1064,13 +1093,8 @@ static bool do_test_display_audio(struct audio_state *state)
 
 	success = false;
 	streak = 0;
-	msec = 0;
-	i = 0;
-	while (!success && msec < AUDIO_TIMEOUT) {
-		ok = chamelium_stream_receive_realtime_audio(state->stream,
-							     &page_count,
-							     &recv, &recv_len);
-		igt_assert(ok);
+	while (!success && state->msec < AUDIO_TIMEOUT) {
+		audio_state_receive(state, &recv, &recv_len);
 
 		memcpy(&buf[buf_len], recv, recv_len * sizeof(int32_t));
 		buf_len += recv_len;

@@ -1079,13 +1103,7 @@ static bool do_test_display_audio(struct audio_state *state)
 			continue;
 		igt_assert(buf_len == buf_cap);
 
-		if (state->dump_fd >= 0) {
-			buf_size = buf_len * sizeof(int32_t);
-			igt_assert(write(state->dump_fd, buf, buf_size) == buf_size);
-		}
-
-		msec = i * channel_len / (double) state->capture.rate * 1000;
-		igt_debug("Detecting audio signal, t=%d msec\n", msec);
+		igt_debug("Detecting audio signal, t=%d msec\n", state->msec);
 
 		for (j = 0; j < state->playback.channels; j++) {
 			capture_chan = state->channel_mapping[j];

@@ -1107,7 +1125,6 @@ static bool do_test_display_audio(struct audio_state *state)
 		}
 
 		buf_len = 0;
-		i++;
 
 		success = streak == MIN_STREAK * state->playback.channels;
 	}

index 71489e5df9d7..f934a5bee74a 100644
@@ -1009,7 +1009,7 @@ static void audio_state_stop(struct audio_state *state, bool success)
 }
 
 static int
-audio_output_callback(void *data, void *buffer, int samples)
+audio_output_frequencies_callback(void *data, void *buffer, int samples)
 {
 	struct audio_state *state = data;
 

@@ -1030,7 +1030,7 @@ audio_output_callback(void *data, void *buffer, int samples)
 	return state->run ? 0 : -1;
 }
 
-static bool do_test_display_audio(struct audio_state *state)
+static bool test_audio_frequencies(struct audio_state *state)
 {
 	int freq, step;
 	int32_t *recv, *buf;

@@ -1064,7 +1064,8 @@ static bool do_test_display_audio(struct audio_state *state)
 	}
 	audio_signal_synthesize(state->signal);
 
-	alsa_register_output_callback(state->alsa, audio_output_callback, state,
+	alsa_register_output_callback(state->alsa,
+				      audio_output_frequencies_callback, state,
 				      PLAYBACK_SAMPLES);
 
 	audio_state_start(state);

@@ -1139,8 +1140,8 @@ static bool do_test_display_audio(struct audio_state *state)
 	return success;
 }
 
-static bool test_audio_configuration(struct alsa *alsa, snd_pcm_format_t format,
-				     int channels, int sampling_rate)
+static bool check_audio_configuration(struct alsa *alsa, snd_pcm_format_t format,
+				      int channels, int sampling_rate)
 {
 	if (!alsa_test_output_configuration(alsa, format, channels,
 					    sampling_rate)) {

@@ -1225,15 +1226,15 @@ test_display_audio(data_t *data, struct chamelium_port *port,
 			channels = PLAYBACK_CHANNELS;
 			sampling_rate = test_sampling_rates[i];
 
-			if (!test_audio_configuration(alsa, format, channels,
-						      sampling_rate))
+			if (!check_audio_configuration(alsa, format, channels,
+						       sampling_rate))
 				continue;
 
 			run = true;
 
 			audio_state_init(&state, data, alsa, port,
 					 format, channels, sampling_rate);
-			success &= do_test_display_audio(&state);
+			success &= test_audio_frequencies(&state);
 			audio_state_fini(&state);
 
 			alsa_close_output(alsa);

index f934a5bee74a..8c63cee5e2e4 100644
@@ -1221,7 +1221,9 @@ test_display_audio(data_t *data, struct chamelium_port *port,
 			ret = alsa_open_output(alsa, audio_device);
 			igt_assert(ret >= 0);
 
-			/* TODO: playback on all 8 available channels */
+			/* TODO: playback on all 8 available channels (this
+			 * isn't supported by Chamelium devices yet, see
+			 * https://crbug.com/950917) */
 			format = test_formats[j];
 			channels = PLAYBACK_CHANNELS;
 			sampling_rate = test_sampling_rates[i];

index ea2d83a5cad0..be665b03116f 100644
@@ -304,51 +304,6 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 	audio_sanity_check(buffer, signal->channels * samples);
 }
 
-void audio_signal_fill_s16_le(struct audio_signal *signal, int16_t *buffer,
-			      size_t samples)
-{
-	double *tmp;
-	size_t i;
-
-	tmp = malloc(sizeof(double) * signal->channels * samples);
-	audio_signal_fill(signal, tmp, samples);
-
-	for (i = 0; i < signal->channels * samples; ++i)
-		buffer[i] = INT16_MAX * tmp[i];
-
-	free(tmp);
-}
-
-void audio_signal_fill_s24_le(struct audio_signal *signal, int32_t *buffer,
-			      size_t samples)
-{
-	double *tmp;
-	size_t i;
-
-	tmp = malloc(sizeof(double) * signal->channels * samples);
-	audio_signal_fill(signal, tmp, samples);
-
-	for (i = 0; i < signal->channels * samples; ++i)
-		buffer[i] = 0x7FFFFF * tmp[i];
-
-	free(tmp);
-}
-
-void audio_signal_fill_s32_le(struct audio_signal *signal, int32_t *buffer,
-			      size_t samples)
-{
-	double *tmp;
-	size_t i;
-
-	tmp = malloc(sizeof(double) * signal->channels * samples);
-	audio_signal_fill(signal, tmp, samples);
-
-	for (i = 0; i < signal->channels * samples; ++i)
-		buffer[i] = INT32_MAX * tmp[i];
-
-	free(tmp);
-}
-
 /**
  * Checks that frequencies specified in signal, and only those, are included
  * in the input data.

@@ -508,6 +463,48 @@ size_t audio_extract_channel_s32_le(double *dst, size_t dst_cap,
 	return dst_len;
 }
 
+static void audio_convert_to_s16_le(int16_t *dst, double *src, size_t len)
+{
+	size_t i;
+
+	for (i = 0; i < len; ++i)
+		dst[i] = INT16_MAX * src[i];
+}
+
+static void audio_convert_to_s24_le(int32_t *dst, double *src, size_t len)
+{
+	size_t i;
+
+	for (i = 0; i < len; ++i)
+		dst[i] = 0x7FFFFF * src[i];
+}
+
+static void audio_convert_to_s32_le(int32_t *dst, double *src, size_t len)
+{
+	size_t i;
+
+	for (i = 0; i < len; ++i)
+		dst[i] = INT32_MAX * src[i];
+}
+
+void audio_convert_to(void *dst, double *src, size_t len,
+		      snd_pcm_format_t format)
+{
+	switch (format) {
+	case SND_PCM_FORMAT_S16_LE:
+		audio_convert_to_s16_le(dst, src, len);
+		break;
+	case SND_PCM_FORMAT_S24_LE:
+		audio_convert_to_s24_le(dst, src, len);
+		break;
+	case SND_PCM_FORMAT_S32_LE:
+		audio_convert_to_s32_le(dst, src, len);
+		break;
+	default:
+		assert(false); /* unreachable */
+	}
+}
+
 #define RIFF_TAG "RIFF"
 #define WAVE_TAG "WAVE"
 #define FMT_TAG "fmt "

index c8de70871faa..5c910c27304d 100644
@@ -32,6 +32,8 @@
 #include <stdbool.h>
 #include <stdint.h>
 
+#include <alsa/asoundlib.h>
+
 struct audio_signal;
 
 struct audio_signal *audio_signal_init(int channels, int sampling_rate);

@@ -41,18 +43,14 @@ int audio_signal_add_frequency(struct audio_signal *signal, int frequency,
 void audio_signal_synthesize(struct audio_signal *signal);
 void audio_signal_reset(struct audio_signal *signal);
 void audio_signal_fill(struct audio_signal *signal, double *buffer,
-		       size_t buffer_len);
-void audio_signal_fill_s16_le(struct audio_signal *signal, int16_t *buffer,
-			      size_t buffer_len);
-void audio_signal_fill_s24_le(struct audio_signal *signal, int32_t *buffer,
-			      size_t buffer_len);
-void audio_signal_fill_s32_le(struct audio_signal *signal, int32_t *buffer,
-			      size_t buffer_len);
+		       size_t samples);
 bool audio_signal_detect(struct audio_signal *signal, int sampling_rate,
 			 int channel, const double *samples, size_t samples_len);
 size_t audio_extract_channel_s32_le(double *dst, size_t dst_cap,
 				    int32_t *src, size_t src_len,
 				    int n_channels, int channel);
+void audio_convert_to(void *dst, double *src, size_t len,
+		      snd_pcm_format_t format);
 int audio_create_wav_file_s32_le(const char *qualifier, uint32_t sample_rate,
 				 uint16_t channels, char **path);
 

index 8c63cee5e2e4..065cb1676a3b 100644
@@ -1012,20 +1012,14 @@ static int
 audio_output_frequencies_callback(void *data, void *buffer, int samples)
 {
 	struct audio_state *state = data;
-
-	switch (state->playback.format) {
-	case SND_PCM_FORMAT_S16_LE:
-		audio_signal_fill_s16_le(state->signal, buffer, samples);
-		break;
-	case SND_PCM_FORMAT_S24_LE:
-		audio_signal_fill_s24_le(state->signal, buffer, samples);
-		break;
-	case SND_PCM_FORMAT_S32_LE:
-		audio_signal_fill_s32_le(state->signal, buffer, samples);
-		break;
-	default:
-		assert(false); /* unreachable */
-	}
+	double *tmp;
+	size_t len;
+
+	len = samples * state->playback.channels;
+	tmp = malloc(len * sizeof(double));
+	audio_signal_fill(state->signal, tmp, samples);
+	audio_convert_to(buffer, tmp, len, state->playback.format);
+	free(tmp);
 
 	return state->run ? 0 : -1;
 }

index 065cb1676a3b..40ca93687c20 100644
@@ -824,6 +824,7 @@ struct audio_state {
 		int rate;
 	} playback, capture;
 
+	char *name;
 	struct audio_signal *signal;
 	int channel_mapping[CHAMELIUM_MAX_AUDIO_CHANNELS];
 

@@ -861,6 +862,7 @@ static void audio_state_init(struct audio_state *state, data_t *data,
 static void audio_state_fini(struct audio_state *state)
 {
 	chamelium_stream_deinit(state->stream);
+	free(state->name);
 }
 
 static void *run_audio_thread(void *data)

@@ -871,7 +873,7 @@ static void *run_audio_thread(void *data)
 	return NULL;
 }
 
-static void audio_state_start(struct audio_state *state)
+static void audio_state_start(struct audio_state *state, const char *name)
 {
 	int ret;
 	bool ok;

@@ -879,12 +881,14 @@ static void audio_state_start(struct audio_state *state)
 	enum chamelium_stream_realtime_mode stream_mode;
 	char dump_suffix[64];
 
+	free(state->name);
+	state->name = strdup(name);
 	state->recv_pages = 0;
 	state->msec = 0;
 
-	igt_debug("Starting test with playback format %s, sampling rate %d Hz "
-		  "and %d channels\n",
-		  snd_pcm_format_name(state->playback.format),
+	igt_debug("Starting %s test with playback format %s, "
+		  "sampling rate %d Hz and %d channels\n",
+		  name, snd_pcm_format_name(state->playback.format),
 		  state->playback.rate, state->playback.channels);
 
 	chamelium_start_capturing_audio(state->chamelium, state->port, false);

@@ -930,8 +934,8 @@ static void audio_state_start(struct audio_state *state)
 
 	if (igt_frame_dump_is_enabled()) {
 		snprintf(dump_suffix, sizeof(dump_suffix),
-			 "capture-%s-%dch-%dHz",
-			 snd_pcm_format_name(state->playback.format),
+			 "capture-%s-%s-%dch-%dHz",
+			 name, snd_pcm_format_name(state->playback.format),
 			 state->playback.channels, state->playback.rate);
 
 		state->dump_fd = audio_create_wav_file_s32_le(dump_suffix,

@@ -1001,9 +1005,9 @@ static void audio_state_stop(struct audio_state *state, bool success)
 		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),
+	igt_debug("Audio %s test result for format %s, sampling rate %d Hz "
+		  "and %d channels: %s\n",
+		  state->name, snd_pcm_format_name(state->playback.format),
 		  state->playback.rate, state->playback.channels,
 		  success ? "ALL GREEN" : "FAILED");
 }

@@ -1062,7 +1066,7 @@ static bool test_audio_frequencies(struct audio_state *state)
 				      audio_output_frequencies_callback, state,
 				      PLAYBACK_SAMPLES);
 
-	audio_state_start(state);
+	audio_state_start(state, "frequencies");
 
 	igt_assert(state->capture.rate == state->playback.rate);
 

index be665b03116f..08c0fb6af0db 100644
@@ -445,7 +445,13 @@ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate,
 }
 
 /**
- * Extracts a single channel from a multi-channel S32_LE input buffer.
+ * audio_extract_channel_s32_le: extracts a single channel from a multi-channel
+ * S32_LE input buffer.
+ *
+ * If dst_cap is zero, no copy is performed. This can be used to compute the
+ * minimum required capacity.
+ *
+ * Returns: the number of samples extracted.
  */
 size_t audio_extract_channel_s32_le(double *dst, size_t dst_cap,
 				    int32_t *src, size_t src_len,

@@ -456,6 +462,9 @@ size_t audio_extract_channel_s32_le(double *dst, size_t dst_cap,
 	igt_assert(channel < n_channels);
 	igt_assert(src_len % n_channels == 0);
 	dst_len = src_len / n_channels;
+	if (dst_cap == 0)
+		return dst_len;
+
 	igt_assert(dst_len <= dst_cap);
 	for (i = 0; i < dst_len; i++)
 		dst[i] = (double) src[i * n_channels + channel] / INT32_MAX;

index 40ca93687c20..451a616f1a2e 100644
@@ -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] */
+#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);

index 451a616f1a2e..49c3de0b7beb 100644
@@ -773,7 +773,8 @@ test_display_frame_dump(data_t *data, struct chamelium_port *port)
 #define MIN_STREAK 3
 
 #define FLATLINE_AMPLITUDE 0.9 /* normalized, ie. in [0, 1] */
-#define FLATLINE_ACCURACY 0.001 /* ± 0.1% of the full amplitude */
+#define FLATLINE_AMPLITUDE_ACCURACY 0.001 /* ± 0.1 % of the full amplitude */
+#define FLATLINE_ALIGN_ACCURACY 0 /* number of samples */
 
 /* TODO: enable >48KHz rates, these are not reliable */
 static int test_sampling_rates[] = {

@@ -828,7 +829,7 @@ struct audio_state {
 	} playback, capture;
 
 	char *name;
-	struct audio_signal *signal;
+	struct audio_signal *signal; /* for frequencies test only */
 	int channel_mapping[CHAMELIUM_MAX_AUDIO_CHANNELS];
 
 	size_t recv_pages;

@@ -839,6 +840,7 @@ struct audio_state {
 
 	pthread_t thread;
 	atomic_bool run;
+	atomic_bool positive; /* for pulse test only */
 };
 
 static void audio_state_init(struct audio_state *state, data_t *data,

@@ -1151,16 +1153,16 @@ static int audio_output_flatline_callback(void *data, void *buffer,
 	len = samples * state->playback.channels;
 	tmp = malloc(len * sizeof(double));
 	for (i = 0; i < len; i++)
-		tmp[i] = FLATLINE_AMPLITUDE;
+		tmp[i] = (state->positive ? 1 : -1) * 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)
+static bool detect_flatline_amplitude(double *buf, size_t buf_len, bool pos)
 {
-	double min, max;
+	double expected, min, max;
 	size_t i;
 	bool ok;
 

@@ -1172,34 +1174,63 @@ static bool detect_flatline_amplitude(double *buf, size_t buf_len)
 			max = buf[i];
 	}
 
-	ok = (min >= FLATLINE_AMPLITUDE - FLATLINE_ACCURACY &&
-	      max <= FLATLINE_AMPLITUDE + FLATLINE_ACCURACY);
+	expected = (pos ? 1 : -1) * FLATLINE_AMPLITUDE;
+	ok = (min >= expected - FLATLINE_AMPLITUDE_ACCURACY &&
+	      max <= expected + FLATLINE_AMPLITUDE_ACCURACY);
 	if (ok)
-		igt_debug("Flatline detected\n");
+		igt_debug("Flatline wave amplitude detected\n");
 	else
-		igt_debug("Flatline not detected (min=%f, max=%f)\n",
+		igt_debug("Flatline amplitude not detected (min=%f, max=%f)\n",
 			  min, max);
 	return ok;
 }
 
+static ssize_t detect_falling_edge(double *buf, size_t buf_len)
+{
+	size_t i;
+
+	for (i = 0; i < buf_len; i++) {
+		if (buf[i] < 0)
+			return i;
+	}
+
+	return -1;
+}
+
+/** test_audio_flatline:
+ *
+ * Send a constant value (one positive, then a negative one) and check that:
+ *
+ * - The amplitude of the flatline is correct
+ * - All channels switch from a positive signal to a negative one at the same
+ *   time (ie. all channels are aligned)
+ */
 static bool test_audio_flatline(struct audio_state *state)
 {
-	bool success;
+	bool success, amp_success, align_success;
 	int32_t *recv;
 	size_t recv_len, i, channel_len;
+	ssize_t j;
 	int streak, capture_chan;
 	double *channel;
+	int falling_edges[CHAMELIUM_MAX_AUDIO_CHANNELS];
 
 	alsa_register_output_callback(state->alsa,
 				      audio_output_flatline_callback, state,
 				      PLAYBACK_SAMPLES);
 
+	/* Start by sending a positive signal */
+	state->positive = true;
+
 	audio_state_start(state, "flatline");
 
+	for (i = 0; i < state->playback.channels; i++)
+		falling_edges[i] = -1;
+
 	recv = NULL;
 	recv_len = 0;
-	success = false;
-	while (!success && state->msec < AUDIO_TIMEOUT) {
+	amp_success = false;
+	while (!amp_success && state->msec < AUDIO_TIMEOUT) {
 		audio_state_receive(state, &recv, &recv_len);
 
 		igt_debug("Detecting audio signal, t=%d msec\n", state->msec);

@@ -1220,17 +1251,58 @@ static bool test_audio_flatline(struct audio_state *state)
 						     state->capture.channels,
 						     capture_chan);
 
-			if (detect_flatline_amplitude(channel, channel_len))
+			/* Check whether the amplitude is fine */
+			if (detect_flatline_amplitude(channel, channel_len,
+						      state->positive))
 				streak++;
 			else
 				streak = 0;
 
+			/* If we're now sending a negative signal, detect the
+			 * falling edge */
+			j = detect_falling_edge(channel, channel_len);
+			if (!state->positive && j >= 0) {
+				falling_edges[i] = recv_len * state->recv_pages
+						   + j;
+			}
+
 			free(channel);
 		}
 
-		success = streak == MIN_STREAK * state->playback.channels;
+		amp_success = streak == MIN_STREAK * state->playback.channels;
+
+		if (amp_success && state->positive) {
+			/* Switch to a negative signal after we've detected the
+			 * positive one. */
+			state->positive = false;
+			amp_success = false;
+			streak = 0;
+			igt_debug("Switching to negative square wave\n");
+		}
+	}
+
+	/* Check alignment between all channels by comparing the index of the
+	 * falling edge. */
+	align_success = true;
+	for (i = 0; i < state->playback.channels; i++) {
+		if (falling_edges[i] < 0) {
+			igt_debug("Falling edge not detected for channel %zu\n",
+				  i);
+			align_success = false;
+			continue;
+		}
+
+		if (abs(falling_edges[0] - falling_edges[i]) >
+		    FLATLINE_ALIGN_ACCURACY) {
+			igt_debug("Channel alignment mismatch: "
+				  "channel 0 has a falling edge at index %d "
+				  "while channel %zu has index %d\n",
+				  falling_edges[0], i, falling_edges[i]);
+			align_success = false;
+		}
 	}
 
+	success = amp_success && align_success;
 	audio_state_stop(state, success);
 
 	free(recv);