lib/igt_audio: fix synthesized signal amplitude

1/2

---
 lib/igt_audio.c | 33 ++++++++++++++++++++++++++++-----
 1 file changed, 28 insertions(+), 5 deletions(-)
index 90d16fe4bd11..f6c8e399e345 100644
@@ -36,6 +36,8 @@
 #include "igt_core.h"
 
 #define FREQS_MAX 64
+#define CHANNELS_MAX 8
+#define SYNTHESIZE_AMPLITUDE 0.9
 
 /**
  * SECTION:igt_audio

@@ -77,6 +79,8 @@ struct audio_signal *audio_signal_init(int channels, int sampling_rate)
 {
 	struct audio_signal *signal;
 
+	igt_assert(channels <= CHANNELS_MAX);
+
 	signal = malloc(sizeof(struct audio_signal));
 	memset(signal, 0, sizeof(struct audio_signal));
 

@@ -156,7 +160,7 @@ void audio_signal_synthesize(struct audio_signal *signal)
 
 		for (j = 0; j < period_len; j++) {
 			value = 2.0 * M_PI * freq / signal->sampling_rate * j;
-			value = sin(value) / signal->freqs_count;
+			value = sin(value) * SYNTHESIZE_AMPLITUDE;
 
 			period[j] = value;
 		}

@@ -195,6 +199,20 @@ void audio_signal_reset(struct audio_signal *signal)
 	signal->freqs_count = 0;
 }
 
+static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)
+{
+	size_t n, i;
+	struct audio_signal_freq *freq;
+
+	for (i = 0; i < signal->freqs_count; i++) {
+		freq = &signal->freqs[i];
+		if (freq->channel < 0 || freq->channel == channel)
+			n++;
+	}
+
+	return n;
+}
+
 /**
  * audio_signal_fill:
  * @signal: The target signal structure

@@ -208,14 +226,18 @@ void audio_signal_reset(struct audio_signal *signal)
 void audio_signal_fill(struct audio_signal *signal, double *buffer,
 		       size_t samples)
 {
-	double *destination, *source;
+	double *dst, *src;
 	struct audio_signal_freq *freq;
 	int total;
 	int count;
 	int i, j, k;
+	size_t freqs_per_channel[CHANNELS_MAX];
 
 	memset(buffer, 0, sizeof(double) * signal->channels * samples);
 
+	for (i = 0; i < signal->channels; i++)
+		freqs_per_channel[i] = audio_signal_count_freqs(signal, i);
+
 	for (i = 0; i < signal->freqs_count; i++) {
 		freq = &signal->freqs[i];
 		total = 0;

@@ -223,8 +245,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 		igt_assert(freq->period);
 
 		while (total < samples) {
-			source = freq->period + freq->offset;
-			destination = buffer + total * signal->channels;
+			src = freq->period + freq->offset;
+			dst = buffer + total * signal->channels;
 
 			count = freq->period_len - freq->offset;
 			if (count > samples - total)

@@ -238,7 +260,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 					if (freq->channel >= 0 &&
 					    freq->channel != k)
 						continue;
-					destination[j * signal->channels + k] += source[j];
+					dst[j * signal->channels + k] +=
+						src[j] / freqs_per_channel[k];
 				}
 			}
 

2.21.0

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igt-dev mailing list
[email protected]
https://lists.freedesktop.org/mailman/listinfo/igt-dev
---
 lib/igt_audio.c | 30 ++++++++++++++++++++++++++++++
 1 file changed, 30 insertions(+)
index f6c8e399e345..0d7422c6f3ec 100644
@@ -213,6 +213,34 @@ static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)
 	return n;
 }
 
+/** audio_sanity_check:
+ *
+ * Make sure our generated signal is not messed up. In particular, make sure
+ * the maximum reaches a reasonable value but doesn't exceed our
+ * SYNTHESIZE_AMPLITUDE limit. Same for the minimum.
+ *
+ * We want the signal to be powerful enough to be able to hear something. We
+ * want the signal not to reach 1.0 so that we're sure it won't get capped by
+ * the audio card or the receiver.
+ */
+static void audio_sanity_check(double *samples, size_t samples_len)
+{
+	size_t i;
+	double min = 0, max = 0;
+
+	for (i = 0; i < samples_len; i++) {
+		if (samples[i] < min)
+			min = samples[i];
+		if (samples[i] > max)
+			max = samples[i];
+	}
+
+	igt_assert(-SYNTHESIZE_AMPLITUDE <= min);
+	igt_assert(min <= -SYNTHESIZE_AMPLITUDE + 0.2);
+	igt_assert(SYNTHESIZE_AMPLITUDE - 0.2 <= max);
+	igt_assert(max <= SYNTHESIZE_AMPLITUDE);
+}
+
 /**
  * audio_signal_fill:
  * @signal: The target signal structure

@@ -268,6 +296,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 			total += count;
 		}
 	}
+
+	audio_sanity_check(buffer, signal->channels * samples);
 }
 
 void audio_signal_fill_s16_le(struct audio_signal *signal, int16_t *buffer,

2.21.0

_______________________________________________
igt-dev mailing list
[email protected]
https://lists.freedesktop.org/mailman/listinfo/igt-dev
---
 lib/igt_audio.c | 30 ++++++++++++++++++++++++++++++
 1 file changed, 30 insertions(+)
index f6c8e399e345..0d7422c6f3ec 100644
@@ -213,6 +213,34 @@ static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)
 	return n;
 }
 
+/** audio_sanity_check:
+ *
+ * Make sure our generated signal is not messed up. In particular, make sure
+ * the maximum reaches a reasonable value but doesn't exceed our
+ * SYNTHESIZE_AMPLITUDE limit. Same for the minimum.
+ *
+ * We want the signal to be powerful enough to be able to hear something. We
+ * want the signal not to reach 1.0 so that we're sure it won't get capped by
+ * the audio card or the receiver.
+ */
+static void audio_sanity_check(double *samples, size_t samples_len)
+{
+	size_t i;
+	double min = 0, max = 0;
+
+	for (i = 0; i < samples_len; i++) {

+		if (samples[i] < min)
+			min = samples[i];
+		if (samples[i] > max)
+			max = samples[i];
+	}
+
+	igt_assert(-SYNTHESIZE_AMPLITUDE <= min);
+	igt_assert(min <= -SYNTHESIZE_AMPLITUDE + 0.2);
+	igt_assert(SYNTHESIZE_AMPLITUDE - 0.2 <= max);
+	igt_assert(max <= SYNTHESIZE_AMPLITUDE);

+}
+
 /**
  * audio_signal_fill:
  * @signal: The target signal structure

@@ -268,6 +296,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 			total += count;
 		}
 	}
+
+	audio_sanity_check(buffer, signal->channels * samples);
 }
 
 void audio_signal_fill_s16_le(struct audio_signal *signal, int16_t *buffer,

---
 lib/igt_audio.c | 33 ++++++++++++++++++++++++++++-----
 1 file changed, 28 insertions(+), 5 deletions(-)
index 90d16fe4bd11..f6c8e399e345 100644
@@ -36,6 +36,8 @@
 #include "igt_core.h"
 
 #define FREQS_MAX 64
+#define CHANNELS_MAX 8
+#define SYNTHESIZE_AMPLITUDE 0.9
 
 /**
  * SECTION:igt_audio

@@ -77,6 +79,8 @@ struct audio_signal *audio_signal_init(int channels, int sampling_rate)
 {
 	struct audio_signal *signal;
 
+	igt_assert(channels <= CHANNELS_MAX);
+
 	signal = malloc(sizeof(struct audio_signal));
 	memset(signal, 0, sizeof(struct audio_signal));
 

@@ -156,7 +160,7 @@ void audio_signal_synthesize(struct audio_signal *signal)
 
 		for (j = 0; j < period_len; j++) {
 			value = 2.0 * M_PI * freq / signal->sampling_rate * j;
-			value = sin(value) / signal->freqs_count;
+			value = sin(value) * SYNTHESIZE_AMPLITUDE;
 
 			period[j] = value;
 		}

@@ -195,6 +199,20 @@ void audio_signal_reset(struct audio_signal *signal)
 	signal->freqs_count = 0;
 }
 
+static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)
+{
+	size_t n, i;
+	struct audio_signal_freq *freq;
+
+	for (i = 0; i < signal->freqs_count; i++) {
+		freq = &signal->freqs[i];
+		if (freq->channel < 0 || freq->channel == channel)
+			n++;
+	}
+
+	return n;
+}
+
 /**
  * audio_signal_fill:
  * @signal: The target signal structure

@@ -208,14 +226,18 @@ void audio_signal_reset(struct audio_signal *signal)
 void audio_signal_fill(struct audio_signal *signal, double *buffer,
 		       size_t samples)
 {
-	double *destination, *source;
+	double *dst, *src;
 	struct audio_signal_freq *freq;
 	int total;
 	int count;
 	int i, j, k;
+	size_t freqs_per_channel[CHANNELS_MAX];
 
 	memset(buffer, 0, sizeof(double) * signal->channels * samples);
 
+	for (i = 0; i < signal->channels; i++)
+		freqs_per_channel[i] = audio_signal_count_freqs(signal, i);
+
 	for (i = 0; i < signal->freqs_count; i++) {
 		freq = &signal->freqs[i];
 		total = 0;

@@ -223,8 +245,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 		igt_assert(freq->period);
 
 		while (total < samples) {
-			source = freq->period + freq->offset;
-			destination = buffer + total * signal->channels;
+			src = freq->period + freq->offset;
+			dst = buffer + total * signal->channels;
 
 			count = freq->period_len - freq->offset;
 			if (count > samples - total)

@@ -238,7 +260,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
 					if (freq->channel >= 0 &&
 					    freq->channel != k)
 						continue;
-					destination[j * signal->channels + k] += source[j];
+					dst[j * signal->channels + k] +=
+						src[j] / freqs_per_channel[k];
 				}
 			}
 


lib/igt_audio: fix synthesized signal amplitude

Simon Ser

Simon Ser

Martin Peres

Patchwork

Martin Peres

Patchwork