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authorMax Kellermann <max@duempel.org>2012-03-21 19:09:22 +0100
committerMax Kellermann <max@duempel.org>2012-03-21 19:14:05 +0100
commit3dba09f339407716a20054b34e043e47470c645f (patch)
tree7185447ecc68059e6d785415cfcca4460bd9009b /src/output/alsa_output_plugin.c
parent7ebf8e66c9336f93ddf9d747a917f59595fd9dec (diff)
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output/alsa: always receive host byte order samples
Don't use audio_format.reverse_endian.
Diffstat (limited to '')
-rw-r--r--src/output/alsa_output_plugin.c64
1 files changed, 61 insertions, 3 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index 35bea3ce5..c62ad2a46 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/alsa_output_plugin.c
@@ -21,6 +21,8 @@
#include "alsa_output_plugin.h"
#include "output_api.h"
#include "mixer_list.h"
+#include "pcm_buffer.h"
+#include "pcm_byteswap.h"
#include <glib.h>
#include <alsa/asoundlib.h>
@@ -45,6 +47,13 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
struct alsa_data {
struct audio_output base;
+ /**
+ * The buffer used to reverse the byte order.
+ *
+ * @see #reverse_endian
+ */
+ struct pcm_buffer reverse_buffer;
+
/** the configured name of the ALSA device; NULL for the
default device */
char *device;
@@ -52,6 +61,21 @@ struct alsa_data {
/** use memory mapped I/O? */
bool use_mmap;
+ /**
+ * Does ALSA expect samples in reverse byte order? (i.e. not
+ * host byte order)
+ *
+ * This attribute is only valid while the device is open.
+ */
+ bool reverse_endian;
+
+ /**
+ * Which sample format is being sent to the play() method?
+ *
+ * This attribute is only valid while the device is open.
+ */
+ enum sample_format sample_format;
+
/** libasound's buffer_time setting (in microseconds) */
unsigned int buffer_time;
@@ -168,6 +192,23 @@ alsa_finish(struct audio_output *ao)
}
static bool
+alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
+{
+ struct alsa_data *ad = (struct alsa_data *)ao;
+
+ pcm_buffer_init(&ad->reverse_buffer);
+ return true;
+}
+
+static void
+alsa_output_disable(struct audio_output *ao)
+{
+ struct alsa_data *ad = (struct alsa_data *)ao;
+
+ pcm_buffer_deinit(&ad->reverse_buffer);
+}
+
+static bool
alsa_test_default_device(void)
{
snd_pcm_t *handle;
@@ -288,13 +329,18 @@ alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
static int
alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
struct audio_format *audio_format,
+ bool *reverse_endian_r,
enum sample_format sample_format)
{
+ *reverse_endian_r = false;
+
int err = alsa_output_try_format(pcm, hwparams, audio_format,
sample_format);
- if (err == -EINVAL)
+ if (err == -EINVAL) {
+ *reverse_endian_r = true;
err = alsa_output_try_reverse(pcm, hwparams, audio_format,
sample_format);
+ }
return err;
}
@@ -304,11 +350,13 @@ alsa_output_try_format_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
*/
static int
alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
- struct audio_format *audio_format)
+ struct audio_format *audio_format,
+ bool *reverse_endian_r)
{
/* try the input format first */
int err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+ reverse_endian_r,
audio_format->format);
if (err != -EINVAL)
return err;
@@ -329,6 +377,7 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
continue;
err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+ reverse_endian_r,
probe_formats[i]);
if (err != -EINVAL)
return err;
@@ -387,7 +436,8 @@ configure_hw:
ad->writei = snd_pcm_writei;
}
- err = alsa_output_setup_format(ad->pcm, hwparams, audio_format);
+ err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
+ &ad->reverse_endian);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support format %s: %s",
@@ -397,6 +447,8 @@ configure_hw:
return false;
}
+ ad->sample_format = audio_format->format;
+
err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
&channels);
if (err < 0) {
@@ -660,6 +712,10 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
{
struct alsa_data *ad = (struct alsa_data *)ao;
+ if (ad->reverse_endian)
+ chunk = pcm_byteswap(&ad->reverse_buffer, ad->sample_format,
+ chunk, size);
+
size /= ad->frame_size;
while (true) {
@@ -684,6 +740,8 @@ const struct audio_output_plugin alsa_output_plugin = {
.test_default_device = alsa_test_default_device,
.init = alsa_init,
.finish = alsa_finish,
+ .enable = alsa_output_enable,
+ .disable = alsa_output_disable,
.open = alsa_open,
.play = alsa_play,
.drain = alsa_drain,