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

Simon Ser
2019-05-24T15:03:56Z

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