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-rw-r--r--src/output/alsa_output_plugin.c41
1 files changed, 10 insertions, 31 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index 00312c435..7714e8a66 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/alsa_output_plugin.c
@@ -21,8 +21,7 @@
#include "alsa_output_plugin.h"
#include "output_api.h"
#include "mixer_list.h"
-#include "pcm_buffer.h"
-#include "pcm_byteswap.h"
+#include "pcm_export.h"
#include <glib.h>
#include <alsa/asoundlib.h>
@@ -47,12 +46,7 @@ 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;
+ struct pcm_export_state export;
/** the configured name of the ALSA device; NULL for the
default device */
@@ -61,21 +55,6 @@ 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;
@@ -196,7 +175,7 @@ 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);
+ pcm_export_init(&ad->export);
return true;
}
@@ -205,7 +184,7 @@ alsa_output_disable(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
- pcm_buffer_deinit(&ad->reverse_buffer);
+ pcm_export_deinit(&ad->export);
}
static bool
@@ -434,8 +413,9 @@ configure_hw:
ad->writei = snd_pcm_writei;
}
+ bool reverse_endian;
err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
- &ad->reverse_endian);
+ &reverse_endian);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support format %s: %s",
@@ -445,8 +425,6 @@ 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) {
@@ -579,6 +557,9 @@ configure_hw:
ad->period_frames = alsa_period_size;
ad->period_position = 0;
+ pcm_export_open(&ad->export, audio_format->format,
+ reverse_endian);
+
return true;
error:
@@ -710,9 +691,7 @@ 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);
+ chunk = pcm_export(&ad->export, chunk, size, &size);
size /= ad->frame_size;