output/{alsa,oss}: move endian code to new library pcm_export

2 files changed, 23 insertions, 61 deletions

diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index 00312c43..7714e8a6 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;
 
diff --git a/src/output/oss_output_plugin.c b/src/output/oss_output_plugin.c
index ea15022d..1b09c4b5 100644
--- a/src/output/oss_output_plugin.c
+++ b/src/output/oss_output_plugin.c
@@ -52,20 +52,14 @@
 #endif
 
 #ifdef AFMT_S24_PACKED
-#include "pcm_buffer.h"
-#include "pcm_byteswap.h"
+#include "pcm_export.h"
 #endif
 
 struct oss_data {
 	struct audio_output base;
 
 #ifdef AFMT_S24_PACKED
-	/**
-	 * The buffer used to reverse the byte order.
-	 *
-	 * @see #reverse_endian
-	 */
-	struct pcm_buffer reverse_buffer;
+	struct pcm_export_state export;
 #endif
 
 	int fd;
@@ -76,16 +70,6 @@ struct oss_data {
 	 * the device after cancel().
 	 */
 	struct audio_format audio_format;
-
-#ifdef AFMT_S24_PACKED
-	/**
-	 * Does OSS 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;
-#endif
 };
 
 /**
@@ -252,7 +236,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
 {
 	struct oss_data *od = (struct oss_data *)ao;
 
-	pcm_buffer_init(&od->reverse_buffer);
+	pcm_export_init(&od->export);
 	return true;
 }
 
@@ -261,7 +245,7 @@ oss_output_disable(struct audio_output *ao)
 {
 	struct oss_data *od = (struct oss_data *)ao;
 
-	pcm_buffer_deinit(&od->reverse_buffer);
+	pcm_export_deinit(&od->export);
 }
 
 #endif
@@ -517,7 +501,7 @@ sample_format_from_oss(int format)
 static bool
 oss_setup_sample_format(int fd, struct audio_format *audio_format,
 #ifdef AFMT_S24_PACKED
-			bool *reverse_endian_r,
+			struct pcm_export_state *export,
 #endif
 			GError **error_r)
 {
@@ -537,8 +521,9 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 		audio_format->format = mpd_format;
 
 #ifdef AFMT_S24_PACKED
-		*reverse_endian_r = oss_format == AFMT_S24_PACKED &&
-			G_BYTE_ORDER != G_LITTLE_ENDIAN;
+		pcm_export_open(export, mpd_format,
+				oss_format == AFMT_S24_PACKED &&
+				G_BYTE_ORDER != G_LITTLE_ENDIAN);
 #endif
 		return true;
 
@@ -583,8 +568,9 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 			audio_format->format = mpd_format;
 
 #ifdef AFMT_S24_PACKED
-			*reverse_endian_r = oss_format == AFMT_S24_PACKED &&
-				G_BYTE_ORDER != G_LITTLE_ENDIAN;
+			pcm_export_open(export, mpd_format,
+					oss_format == AFMT_S24_PACKED &&
+					G_BYTE_ORDER != G_LITTLE_ENDIAN);
 #endif
 			return true;
 
@@ -611,7 +597,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
 		oss_setup_sample_rate(od->fd, audio_format, error_r) &&
 		oss_setup_sample_format(od->fd, audio_format,
 #ifdef AFMT_S24_PACKED
-					&od->reverse_endian,
+					&od->export,
 #endif
 					error_r);
 }
@@ -726,10 +712,7 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		return 0;
 
 #ifdef AFMT_S24_PACKED
-	if (od->reverse_endian)
-		chunk = pcm_byteswap(&od->reverse_buffer,
-				     od->audio_format.format,
-				     chunk, size);
+	chunk = pcm_export(&od->export, chunk, size, &size);
 #endif
 
 	while (true) {