44  * To make sure the audio signal doesn't contain noise, #audio_signal_detect
83  * @channels: The number of channels to use for the signal
84  * @sampling_rate: The sampling rate to use for the signal
86  * Allocate and initialize an audio signal structure with the given parameters.
88  * Returns: A newly-allocated audio signal structure
92 	struct audio_signal *signal;  in audio_signal_init()  local
97 	signal = calloc(1, sizeof(struct audio_signal));  in audio_signal_init()
98 	signal->sampling_rate = sampling_rate;  in audio_signal_init()
99 	signal->channels = channels;  in audio_signal_init()
100 	return signal;  in audio_signal_init()
105  * @signal: The target signal structure
106  * @frequency: The frequency to add to the signal
110  * Add a frequency to the signal.
114 int audio_signal_add_frequency(struct audio_signal *signal, int frequency,  in audio_signal_add_frequency()  argument
117 	size_t index = signal->freqs_count;  in audio_signal_add_frequency()
121 	igt_assert(channel < signal->channels);  in audio_signal_add_frequency()
125 	if (frequency > signal->sampling_rate / 2) {  in audio_signal_add_frequency()
127 			  "sampling rate\n", frequency, signal->sampling_rate);  in audio_signal_add_frequency()
133 	 * signal generation, instead of recurrent calls to sin().  in audio_signal_add_frequency()
135 	frequency = signal->sampling_rate / (signal->sampling_rate / frequency);  in audio_signal_add_frequency()
140 	freq = &signal->freqs[index];  in audio_signal_add_frequency()
145 	signal->freqs_count++;  in audio_signal_add_frequency()
152  * @signal: The target signal structure
154  * Synthesize the data tables for the audio signal, that can later be used
156  * freed with a call to audio_signal_clean when the signal is no longer used.
158 void audio_signal_synthesize(struct audio_signal *signal)  in audio_signal_synthesize()  argument
166 	for (i = 0; i < signal->freqs_count; i++) {  in audio_signal_synthesize()
167 		freq = signal->freqs[i].freq;  in audio_signal_synthesize()
168 		period_len = signal->sampling_rate / freq;  in audio_signal_synthesize()
173 			value = 2.0 * M_PI * freq / signal->sampling_rate * j;  in audio_signal_synthesize()
179 		signal->freqs[i].period = period;  in audio_signal_synthesize()
180 		signal->freqs[i].period_len = period_len;  in audio_signal_synthesize()
187  * Release the signal.
189 void audio_signal_fini(struct audio_signal *signal)  in audio_signal_fini()  argument
191 	audio_signal_reset(signal);  in audio_signal_fini()
192 	free(signal);  in audio_signal_fini()
197  * @signal: The target signal structure
202 void audio_signal_reset(struct audio_signal *signal)  in audio_signal_reset()  argument
206 	for (i = 0; i < signal->freqs_count; i++) {  in audio_signal_reset()
207 		free(signal->freqs[i].period);  in audio_signal_reset()
210 	signal->freqs_count = 0;  in audio_signal_reset()
213 static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)  in audio_signal_count_freqs()  argument
219 	for (i = 0; i < signal->freqs_count; i++) {  in audio_signal_count_freqs()
220 		freq = &signal->freqs[i];  in audio_signal_count_freqs()
230  * Make sure our generated signal is not messed up. In particular, make sure
234  * We want the signal to be powerful enough to be able to hear something. We
235  * want the signal not to reach 1.0 so that we're sure it won't get capped by
258  * @signal: The target signal structure
263  * signal data (in interleaved double format), at the requested sampling rate
268 void audio_signal_fill(struct audio_signal *signal, double *buffer,  in audio_signal_fill()  argument
278 	memset(buffer, 0, sizeof(double) * signal->channels * samples);  in audio_signal_fill()
280 	for (i = 0; i < signal->channels; i++) {  in audio_signal_fill()
281 		freqs_per_channel[i] = audio_signal_count_freqs(signal, i);  in audio_signal_fill()
285 	for (i = 0; i < signal->freqs_count; i++) {  in audio_signal_fill()
286 		freq = &signal->freqs[i];  in audio_signal_fill()
293 			dst = buffer + total * signal->channels;  in audio_signal_fill()
303 				for (k = 0; k < signal->channels; k++) {  in audio_signal_fill()
307 					dst[j * signal->channels + k] +=  in audio_signal_fill()
316 	audio_sanity_check(buffer, signal->channels * samples);  in audio_signal_fill()
326  * Checks that frequencies specified in signal, and only those, are included
332 bool audio_signal_detect(struct audio_signal *signal, int sampling_rate,  in audio_signal_detect()  argument
349 	/* Apply a Hann window to the input signal, to reduce frequency leaks  in audio_signal_detect()
350 	 * due to the endpoints of the signal being discontinuous.  in audio_signal_detect()
409 	for (i = 0; i < signal->freqs_count; i++)  in audio_signal_detect()
442 			for (j = 0; j < signal->freqs_count; j++) {  in audio_signal_detect()
443 				if (signal->freqs[j].channel >= 0 &&  in audio_signal_detect()
444 				    signal->freqs[j].channel != channel)  in audio_signal_detect()
447 				if (signal->freqs[j].freq >  in audio_signal_detect()
449 				    signal->freqs[j].freq <  in audio_signal_detect()
460 			if (j == signal->freqs_count) {  in audio_signal_detect()
478 	for (i = 0; i < signal->freqs_count; i++) {  in audio_signal_detect()
479 		if (signal->freqs[i].channel >= 0 &&  in audio_signal_detect()
480 		    signal->freqs[i].channel != channel)  in audio_signal_detect()
485 				  signal->freqs[i].freq);  in audio_signal_detect()