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;