[igt-dev] [PATCH i-g-t 1/5] lib/igt_alsa: support all alsa formats
Simon Ser
simon.ser at intel.com
Fri May 17 16:02:38 UTC 2019
This commit adds a new format argument to alsa_configure_output. This allows
the caller to decide the format used for the audio signal.
This removes the assumption that samples are S16_LE items in igt_alsa. The
alsa function snd_pcm_format_physical_width is used instead of sizeof(short).
This also removes the theorically incorrect assumption that sizeof(short) ==
sizeof(int16_t).
Right now S16_LE is hardcoded in the test. Tests with other formats will be
added in future commits.
Signed-off-by: Simon Ser <simon.ser at intel.com>
---
lib/igt_alsa.c | 23 +++++++++++++----------
lib/igt_alsa.h | 7 ++++---
tests/kms_chamelium.c | 7 ++++---
3 files changed, 21 insertions(+), 16 deletions(-)
diff --git a/lib/igt_alsa.c b/lib/igt_alsa.c
index 28a866e0ee74..2dfb8ccd3ad9 100644
--- a/lib/igt_alsa.c
+++ b/lib/igt_alsa.c
@@ -27,7 +27,6 @@
#include "config.h"
#include <limits.h>
-#include <alsa/asoundlib.h>
#include "igt_alsa.h"
#include "igt_aux.h"
@@ -47,10 +46,11 @@
struct alsa {
snd_pcm_t *output_handles[HANDLES_MAX];
int output_handles_count;
+ snd_pcm_format_t output_format;
int output_sampling_rate;
int output_channels;
- int (*output_callback)(void *data, short *buffer, int samples);
+ int (*output_callback)(void *data, void *buffer, int samples);
void *output_callback_data;
int output_samples_trigger;
};
@@ -342,8 +342,8 @@ bool alsa_test_output_configuration(struct alsa *alsa, int channels,
* Configure the output devices with the configuration specified by @channels
* and @sampling_rate.
*/
-void alsa_configure_output(struct alsa *alsa, int channels,
- int sampling_rate)
+void alsa_configure_output(struct alsa *alsa, snd_pcm_format_t fmt,
+ int channels, int sampling_rate)
{
snd_pcm_t *handle;
int ret;
@@ -354,13 +354,14 @@ void alsa_configure_output(struct alsa *alsa, int channels,
for (i = 0; i < alsa->output_handles_count; i++) {
handle = alsa->output_handles[i];
- ret = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE,
+ ret = snd_pcm_set_params(handle, fmt,
SND_PCM_ACCESS_RW_INTERLEAVED,
channels, sampling_rate,
soft_resample, latency);
igt_assert(ret >= 0);
}
+ alsa->output_format = fmt;
alsa->output_channels = channels;
alsa->output_sampling_rate = sampling_rate;
}
@@ -379,7 +380,7 @@ void alsa_configure_output(struct alsa *alsa, int channels,
* for failure.
*/
void alsa_register_output_callback(struct alsa *alsa,
- int (*callback)(void *data, short *buffer, int samples),
+ int (*callback)(void *data, void *buffer, int samples),
void *callback_data, int samples_trigger)
{
alsa->output_callback = callback;
@@ -402,12 +403,13 @@ void alsa_register_output_callback(struct alsa *alsa,
int alsa_run(struct alsa *alsa, int duration_ms)
{
snd_pcm_t *handle;
- short *output_buffer = NULL;
+ char *output_buffer = NULL;
int output_limit;
int output_total = 0;
int output_counts[alsa->output_handles_count];
bool output_ready = false;
int output_channels;
+ int bytes_per_sample;
int output_trigger;
bool reached;
int index;
@@ -418,9 +420,10 @@ int alsa_run(struct alsa *alsa, int duration_ms)
output_limit = alsa->output_sampling_rate * duration_ms / 1000;
output_channels = alsa->output_channels;
+ bytes_per_sample = snd_pcm_format_physical_width(alsa->output_format) / 8;
output_trigger = alsa->output_samples_trigger;
- output_buffer = malloc(sizeof(short) * output_channels *
- output_trigger);
+ output_buffer = malloc(output_channels * output_trigger *
+ bytes_per_sample);
do {
reached = true;
@@ -454,7 +457,7 @@ int alsa_run(struct alsa *alsa, int duration_ms)
count = avail < count ? avail : count;
ret = snd_pcm_writei(handle,
- &output_buffer[index],
+ &output_buffer[index * bytes_per_sample],
count);
if (ret < 0) {
ret = snd_pcm_recover(handle,
diff --git a/lib/igt_alsa.h b/lib/igt_alsa.h
index a10985ff777f..46b3568d26fd 100644
--- a/lib/igt_alsa.h
+++ b/lib/igt_alsa.h
@@ -29,6 +29,7 @@
#include "config.h"
+#include <alsa/asoundlib.h>
#include <stdbool.h>
struct alsa;
@@ -39,10 +40,10 @@ int alsa_open_output(struct alsa *alsa, const char *device_name);
void alsa_close_output(struct alsa *alsa);
bool alsa_test_output_configuration(struct alsa *alsa, int channels,
int sampling_rate);
-void alsa_configure_output(struct alsa *alsa, int channels,
- int sampling_rate);
+void alsa_configure_output(struct alsa *alsa, snd_pcm_format_t fmt,
+ int channels, int sampling_rate);
void alsa_register_output_callback(struct alsa *alsa,
- int (*callback)(void *data, short *buffer, int samples),
+ int (*callback)(void *data, void *buffer, int samples),
void *callback_data, int samples_trigger);
int alsa_run(struct alsa *alsa, int duration_ms);
diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
index c8b6b22d7b4a..2b465565418d 100644
--- a/tests/kms_chamelium.c
+++ b/tests/kms_chamelium.c
@@ -801,11 +801,11 @@ struct audio_state {
};
static int
-audio_output_callback(void *data, short *buffer, int frames)
+audio_output_callback(void *data, void *buffer, int samples)
{
struct audio_state *state = data;
- audio_signal_fill_s16_le(state->signal, buffer, frames);
+ audio_signal_fill_s16_le(state->signal, buffer, samples);
return state->run ? 0 : -1;
}
@@ -843,7 +843,8 @@ do_test_display_audio(data_t *data, struct chamelium_port *port,
igt_debug("Testing with playback sampling rate %d Hz and %d channels\n",
playback_rate, playback_channels);
- alsa_configure_output(alsa, playback_channels, playback_rate);
+ alsa_configure_output(alsa, SND_PCM_FORMAT_S16_LE,
+ playback_channels, playback_rate);
chamelium_start_capturing_audio(data->chamelium, port, false);
--
2.21.0
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