[igt-dev] [PATCH i-g-t v2 5/9] lib/igt_audio: introduce audio_convert_to

[Simon Ser <simon.ser@intel.com>]

This function converts normalized doubles into an ALSA PCM format.

Instead of having per-format audio_signal_fill_* functions, we can only have
audio_signal_fill that outputs normalized doubles. Then in ALSA's playback
callback, we can simply use the new audio_convert_to function to fill the
buffer.

This makes the test code simpler and prevents code duplication when another
ALSA playback callback is implemented.

This adds a dependency of igt_audio over ALSA for the PCM format enum, but I
don't think this is a concern, since I don't see the point of using igt_audio
without igt_alsa. If this is an issue, it would always be possible to replace
ALSA's enum with our own in the future.

Signed-off-by: Simon Ser <simon.ser@intel.com>
---
 lib/igt_audio.c       | 87 +++++++++++++++++++++----------------------
 lib/igt_audio.h       | 12 +++---
 tests/kms_chamelium.c | 22 ++++-------
 3 files changed, 55 insertions(+), 66 deletions(-)

diff --git a/lib/igt_audio.c b/lib/igt_audio.c
index 376e04ba6ed6..f2aac0e23a37 100644
--- a/lib/igt_audio.c
+++ b/lib/igt_audio.c
@@ -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 "
diff --git a/lib/igt_audio.h b/lib/igt_audio.h
index c8de70871faa..5c910c27304d 100644
--- a/lib/igt_audio.h
+++ b/lib/igt_audio.h
@@ -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);
 
diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
index 56918a3b43fc..063840721f46 100644
--- a/tests/kms_chamelium.c
+++ b/tests/kms_chamelium.c
@@ -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;
 }
-- 
2.21.0

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