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authorMax Kellermann <max@duempel.org>2011-09-16 23:31:48 +0200
committerMax Kellermann <max@duempel.org>2011-09-19 09:41:21 +0200
commit74617389c88ccf630b8cce4b54d9e2fa5afb2259 (patch)
tree3d05faad2c2b3b2e9e885f2c391a35fe8f06553a /src/output/alsa_output_plugin.c
parentb4a8b8c0d4bf82fc56572c3f8e79108d22d78132 (diff)
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output_plugin: the plugin allocates the audio_output object
Pass audio_output objects around instead of void pointers. This will give some more control to the plugin, and prepares for non-blocking audio outputs.
Diffstat (limited to '')
-rw-r--r--src/output/alsa_output_plugin.c41
1 files changed, 24 insertions, 17 deletions
diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
index fb0498e96..41293272d 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/alsa_output_plugin.c
@@ -43,6 +43,8 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
struct alsa_data {
+ struct audio_output base;
+
/** the configured name of the ALSA device; NULL for the
default device */
char *device;
@@ -143,23 +145,27 @@ alsa_configure(struct alsa_data *ad, const struct config_param *param)
#endif
}
-static void *
-alsa_init(G_GNUC_UNUSED const struct audio_format *audio_format,
- const struct config_param *param,
- G_GNUC_UNUSED GError **error)
+static struct audio_output *
+alsa_init(const struct config_param *param, GError **error_r)
{
struct alsa_data *ad = alsa_data_new();
+ if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
+ g_free(ad);
+ return NULL;
+ }
+
alsa_configure(ad, param);
- return ad;
+ return &ad->base;
}
static void
-alsa_finish(void *data)
+alsa_finish(struct audio_output *ao)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
+ ao_base_finish(&ad->base);
alsa_data_free(ad);
/* free libasound's config cache */
@@ -530,9 +536,9 @@ error:
}
static bool
-alsa_open(void *data, struct audio_format *audio_format, GError **error)
+alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
int err;
bool success;
@@ -594,9 +600,9 @@ alsa_recover(struct alsa_data *ad, int err)
}
static void
-alsa_drain(void *data)
+alsa_drain(struct audio_output *ao)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
return;
@@ -628,9 +634,9 @@ alsa_drain(void *data)
}
static void
-alsa_cancel(void *data)
+alsa_cancel(struct audio_output *ao)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
ad->period_position = 0;
@@ -638,17 +644,18 @@ alsa_cancel(void *data)
}
static void
-alsa_close(void *data)
+alsa_close(struct audio_output *ao)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
snd_pcm_close(ad->pcm);
}
static size_t
-alsa_play(void *data, const void *chunk, size_t size, GError **error)
+alsa_play(struct audio_output *ao, const void *chunk, size_t size,
+ GError **error)
{
- struct alsa_data *ad = data;
+ struct alsa_data *ad = (struct alsa_data *)ao;
size /= ad->frame_size;